Commit 08df0d9a00 ("ASoC: max98090: revert "ASoC: max98090: fix lockdep
warning"") provided a good rationale for removing separate lock for the
SHDN register access. However it restored the lockdep warning during the
system boot. To silence the lockdep warning, mark the mutex taken in the
max98090_shdn_save() function with the lockdep class dedicated for the
runtime DAPM operations: SND_SOC_DAPM_CLASS_RUNTIME. This finally fixes
the following lockdep warning observed on Exynos4412-based Odroid U3
board:
======================================================
WARNING: possible circular locking dependency detected
5.5.0-rc7-next-20200123 #7329 Not tainted
------------------------------------------------------
alsactl/1105 is trying to acquire lock:
ed4f7cf4 (&card->dapm_mutex){+.+.}, at: max98090_shdn_save+0x1c/0x28
but task is already holding lock:
edb8d49c (&card->controls_rwsem){++++}, at: snd_ctl_ioctl+0xcc/0xbb8
which lock already depends on the new lock.
the existing dependency chain (in reverse order) is:
-> #1 (&card->controls_rwsem){++++}:
snd_ctl_add_replace+0x3c/0x84
dapm_create_or_share_kcontrol+0x24c/0x2e0
snd_soc_dapm_new_widgets+0x308/0x594
snd_soc_bind_card+0x834/0xa94
devm_snd_soc_register_card+0x34/0x6c
odroid_audio_probe+0x288/0x34c
platform_drv_probe+0x6c/0xa4
really_probe+0x200/0x48c
driver_probe_device+0x78/0x1f8
bus_for_each_drv+0x74/0xb8
__device_attach+0xd4/0x16c
bus_probe_device+0x88/0x90
deferred_probe_work_func+0x3c/0xd0
process_one_work+0x230/0x7bc
worker_thread+0x44/0x524
kthread+0x130/0x164
ret_from_fork+0x14/0x20
0x0
-> #0 (&card->dapm_mutex){+.+.}:
lock_acquire+0xe8/0x270
__mutex_lock+0x9c/0xb18
mutex_lock_nested+0x1c/0x24
max98090_shdn_save+0x1c/0x28
max98090_put_enum_double+0x20/0x40
snd_ctl_ioctl+0x190/0xbb8
ksys_ioctl+0x484/0xb10
ret_fast_syscall+0x0/0x28
0xbede0564
other info that might help us debug this:
Possible unsafe locking scenario:
CPU0 CPU1
---- ----
lock(&card->controls_rwsem);
lock(&card->dapm_mutex);
lock(&card->controls_rwsem);
lock(&card->dapm_mutex);
*** DEADLOCK ***
1 lock held by alsactl/1105:
#0: edb8d49c (&card->controls_rwsem){++++}, at: snd_ctl_ioctl+0xcc/0xbb8
stack backtrace:
CPU: 2 PID: 1105 Comm: alsactl Not tainted 5.5.0-rc7-next-20200123 #7329
Hardware name: Samsung Exynos (Flattened Device Tree)
[<c01126f0>] (unwind_backtrace) from [<c010e1e8>] (show_stack+0x10/0x14)
[<c010e1e8>] (show_stack) from [<c0b5234c>] (dump_stack+0xb4/0xe0)
[<c0b5234c>] (dump_stack) from [<c018a610>] (check_noncircular+0x1ec/0x208)
[<c018a610>] (check_noncircular) from [<c018ca2c>] (__lock_acquire+0x1210/0x25ec)
[<c018ca2c>] (__lock_acquire) from [<c018e728>] (lock_acquire+0xe8/0x270)
[<c018e728>] (lock_acquire) from [<c0b71928>] (__mutex_lock+0x9c/0xb18)
[<c0b71928>] (__mutex_lock) from [<c0b723c0>] (mutex_lock_nested+0x1c/0x24)
[<c0b723c0>] (mutex_lock_nested) from [<c086097c>] (max98090_shdn_save+0x1c/0x28)
[<c086097c>] (max98090_shdn_save) from [<c08613f8>] (max98090_put_enum_double+0x20/0x40)
[<c08613f8>] (max98090_put_enum_double) from [<c0833f20>] (snd_ctl_ioctl+0x190/0xbb8)
[<c0833f20>] (snd_ctl_ioctl) from [<c02cae14>] (ksys_ioctl+0x484/0xb10)
[<c02cae14>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28)
Exception stack(0xed331fa8 to 0xed331ff0)
...
Fixes: 08df0d9a00 ("ASoC: max98090: revert "ASoC: max98090: fix lockdep warning"")
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200123134046.9769-1-m.szyprowski@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All rtd->dai_link callback functions are controlled by soc_rtd_xxxx(),
and checking rtd->dai_link->ops.
We don't need to have null_snd_soc_ops anymore.
This patch removes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhegl3oz.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soc_rtd_trigger() to make the code easier to read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871rrsmi9j.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soc_rtd_hw_free() to make the code easier to read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736c8mi9n.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soc_rtd_hw_params() to make the code easier to read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874kwomi9r.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soc_rtd_prepare() to make the code easier to read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875zh4mi9v.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soc_rtd_shutdown() to make the code easier to read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e1kmi9z.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soc_rtd_startup() to make the code easier to read
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878sm0mia4.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The code which checks the return value for snd_soc_add_dai_link() call
in soc_tplg_fe_link_create() moved the snd_soc_add_dai_link() call before
link->dobj members initialization.
While it does not affect the latest kernels, the old soc-core.c code
in the stable kernels is affected. The snd_soc_add_dai_link() function uses
the link->dobj.type member to check, if the link structure is valid.
Reorder the link->dobj initialization to make things work again.
It's harmless for the recent code (and the structure should be properly
initialized before other calls anyway).
The problem is in stable linux-5.4.y since version 5.4.11 when the
upstream commit 76d2703649 was applied.
Fixes: 76d2703649 ("ASoC: topology: Check return value for snd_soc_add_dai_link()")
Cc: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20200122190752.3081016-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Both the data and clock should be connected to both the left and right
inputs for DMIC only inputs, add the missing routes.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200122104143.16725-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Lenovo Thinkpad T420s uses the same codec as T420, so apply the
same quirk to enable audio output on a docking station.
Signed-off-by: Peter Große <pegro@friiks.de>
Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MT6660 is a boosted BTL class-D amplifier with V/I sensing.
A built-in DC-DC step-up converter is used to provide efficient
power for class-D amplifier with multi-level class-G operation.
The digital audio interface supports I2S, left-justified,
right-justified, TDM and DSP A/B format for audio in with a data
out used for chip information like voltage sense and current
sense, which are able to be monitored via DATAO through proper
Signed-off-by: Jeff Chang <jeff_chang@richtek.com>
Link: https://lore.kernel.org/r/1579153597-23286-1-git-send-email-richtek.jeff.chang@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 62d5ae4caf ("ASoC: max98090: save and restore SHDN when
changing sensitive registers SHDN bit") uses dapm_mutex to protect SHDN
bit. However, snd_soc_dapm_put_enum_double() in
max98090_dapm_put_enum_double() acquires the dapm_mutex again which
cause a deadlock.
Use snd_soc_dapm_put_enum_double_locked() instead to fix the deadlock.
Fixes: 62d5ae4caf ("ASoC: max98090: save and restore SHDN when changing sensitive registers SHDN bit")
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200117073814.82441-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 2dc98af62c ("ASoC: max98090: fix lockdep warning") introduced
a helpful-less small lock: shdn_lock. Reverts the commit.
Reasons:
1. Lockdep should not be happy by either the original or current code.
From lockdep's point of view, there is a lock inversion anyway.
Let d = dapm_mutex, c = controls_rwsem, s = shdn_lock,
From the reported calling stack: lock acquisition order of
snd_soc_register_card() is: d -> c.
> snd_ctl_add_replace+0x3c/0x84
> dapm_create_or_share_kcontrol+0x24c/0x2e0
> snd_soc_dapm_new_widgets+0x308/0x594
> snd_soc_bind_card+0x80c/0xad4
> devm_snd_soc_register_card+0x34/0x6c
If calling snd_soc_dapm_put_enum_double() in kcontrol's put (e.g.
SOC_DAPM_ENUM_EXT), lock acquisition order is: c -> d. Note that,
snd_soc_dapm_put_enum_double() acquires d.
The possible lock inversion is always there if registering sound card
and putting mixer control happen at the same time. In fact, it never
happens because the control device don't show up to the userspace until
the sound card build success.
Commit 2dc98af62c ("ASoC: max98090: fix lockdep warning") changes the
order to: c -> s -> d. The lock inversion is still there.
2. Commit 62d5ae4caf ("ASoC: max98090: save and restore SHDN when
changing sensitive registers SHDN bit") designed to use dapm_mutex to
protect SHDN bit. Use a separate lock breaks the protection.
DAPM changes SHDN bit automatically when it finds the path. Thus, any
code wants to change the SHDN bit, need to acquire the dapm_mutex first.
> SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_DEVICE_SHUTDOWN,
> M98090_SHDNN_SHIFT, 0, NULL, 0),
Fixes: 2dc98af62c ("ASoC: max98090: fix lockdep warning")
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200117073814.82441-2-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Now, snd_soc_dai_driver::bus_control is used for how to resume.
But, no driver which has bus_control has DAI driver suspend/resume
support.
This patch removes pointless bus_control from ALSA SoC.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnffx7i4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Historically, CPU and Codec were implemented different, but now it is
merged as Component.
ALSA SoC is supporting suspend/resume at DAI and Component level.
The method is like below.
1) Suspend/Resume all CPU DAI if bus-control was 0
2) Suspend/Resume all Component
3) Suspend/Resume all CPU DAI if bus-control was 1
Historically 2) was Codec special operation.
Because CPU and Codec were merged into Component,
CPU suspend/resume has 3 chance to suspend(= 1/2/3), but
Codec suspend/resume has 1 chance (= 2).
Here, DAI side suspend/resume is caring bus-control, but no driver
which is supporting suspend/resume is setting bus-control.
This means 3) was never used.
Here, used parameter for suspend/resume component->dev and dai->dev are
same pointer.
For that reason, we can merge DAI and Component suspend/resume.
One note is that we should use 2), because it is caring BIAS level.
This patch removes 1) and 3).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r1zvx7i8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgkbx7ic.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can swtcih all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tv4rx7ij.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wo9nx7it.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2u3x7iy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhejx7j4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/871rrvym3p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736cbym3x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874kwrym42.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875zh7ym48.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e1nym4e.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878sm3ym4j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a76jym4p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d0bfym53.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87blqzym4w.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the return value checking, that is to align with the code
before adding snd_dmaengine_pcm_refine_runtime_hwparams function.
Otherwise it causes a regression on the HiKey board:
[ 17.721424] hi6210_i2s f7118000.i2s: ASoC: can't open component f7118000.i2s: -6
Fixes: e957204e73 ("ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reported-by: John Stultz <john.stultz@linaro.org>
Link: https://lore.kernel.org/r/1579505286-32085-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Crash happens in snd_soc_dapm_new_dai() when substream->private_data
access is made and substream is NULL here. This is seen for DAIs where
only playback or capture stream is defined. This seems to be happening
for codec2codec DAI link.
Both playback and capture are 0 during soc_new_pcm(). This is probably
happening because cpu_dai and codec_dai are both validated either for
SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE.
Shouldn't be playback = 1 when,
- playback stream is available for codec_dai AND
- capture stream is available for cpu_dai
and vice-versa for capture = 1?
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/1579443563-12287-1-git-send-email-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The initial snd_hda_get_sub_node() can fail on certain
devices (e.g. some Chromebook models using Intel GLK).
The failure rate is very low, but as this is is part of
the probe process, end-user impact is high.
In observed cases, related hardware status registers have
expected values, but the node query still fails. Retrying
the node query does seem to help, so fix the problem by
adding retry logic to the query. This does not impact
non-Intel platforms.
BugLink: https://github.com/thesofproject/linux/issues/1642
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200120160117.29130-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like many other drivers, HD-audio drivers also do PCM buffer
preallocation to assure the buffer pages allocated at the early boot
stage. This step is useful for platforms that may fail to allocate
the PCM hardware buffers -- which is mostly for either large
continuous pages or with the specific DMA mask (like emu10k1).
OTOH, when a buffer is allocated as SG-buffer and the DMA mask is
either 32 or 64 bits, the allocation almost never fails unless it hits
the real OOM situation. In such a case, we don't need the
preallocation inevitably unlike the cases above.
That said, we may drop the preallocation for HD-audio that does
allocate via SG-buffers, and the patch achieves it.
However, there is one caveat: the buffer allocation behavior depends
on CONFIG_SND_DMA_SGBUF, and it falls back to the continuous pages
when it's not set. And, currently this SG buffer allocation is
enabled only on x86 platforms. So, covering those fall-outs, the
patch adjusts CONFIG_SND_HDA_PREALLOC_SIZE depending on the condition,
and keeps the old behavior as-is for non-x86 platforms.
On x86, the kconfig item is no longer adjustable but always set to
zero for disabling the preallocation. You can still enable the
preallocation via procfs interface at any time later, too.
Link: https://lore.kernel.org/r/20200120124423.11862-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the available buffer allocation size for a PCM stream
depends on the preallocated size; when a buffer has been preallocated,
the max buffer size is set to that size, so that application won't
re-allocate too much memory. OTOH, when no preallocation is done,
each substream may allocate arbitrary size of buffers as long as
snd_pcm_hardware.buffer_bytes_max allows -- which can be quite high,
HD-audio sets 1GB there.
It means that the system may consume a high amount of pages for PCM
buffers, and they are pinned and never swapped out. This can lead to
OOM easily.
For avoiding such a situation, this patch adds the upper limit per
card. Each snd_pcm_lib_malloc_pages() and _free_pages() calls are
tracked and it will return an error if the total amount of buffers
goes over the defined upper limit. The default value is set to 32MB,
which should be really large enough for usual operations.
If larger buffers are needed for any specific usage, it can be
adjusted (also dynamically) via snd_pcm.max_alloc_per_card option.
Setting zero there means no chceck is performed, and again, unlimited
amount of buffers are allowed.
Link: https://lore.kernel.org/r/20200120124423.11862-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the recent simplification of HD-audio bus access
helpers caused a regression on the virtual HD-audio device on QEMU
with ARM platforms. The driver got a CORB/RIRB timeout and couldn't
probe any codecs.
The essential difference that caused a problem was the enforced
aligned MMIO accesses by simplification. Since snd-hda-tegra driver
is enabled on ARM, it enables CONFIG_SND_HDA_ALIGNED_MMIO, which makes
the all HD-audio drivers using the aligned MMIO accesses. While this
is mandatory for snd-hda-tegra, it seems that snd-hda-intel on ARM
gets broken by this access pattern.
For addressing the regression, this patch introduces a new flag,
aligned_mmio, to hdac_bus object, and applies the aligned MMIO only
when this flag is set. This change affects only platforms with
CONFIG_SND_HDA_ALIGNED_MMIO set, i.e. mostly only for ARM platforms.
Unfortunately the patch became a big bigger than it should be, just
because the former calls didn't take hdac_bus object in the argument,
hence we had to extend the call patterns.
Fixes: 19abfefd4c ("ALSA: hda: Direct MMIO accesses")
BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1161152
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200120104127.28985-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This became bigger than I have hoped for rc7. But, the only large LOC
is for stm32 fixes that are simple rewriting of register access
helpers, while the rest are all nice and small fixes:
- A few ASoC fixes for the remaining probe error handling bugs
- ALSA sequencer core fix for racy proc file accesses
- Revert the option rename of snd-hda-intel to make compatible again
- Various device-specific fixes
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Merge tag 'sound-5.5-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This became bigger than I have hoped for rc7. But, the only large LOC
is for stm32 fixes that are simple rewriting of register access
helpers, while the rest are all nice and small fixes:
- A few ASoC fixes for the remaining probe error handling bugs
- ALSA sequencer core fix for racy proc file accesses
- Revert the option rename of snd-hda-intel to make compatible again
- Various device-specific fixes"
* tag 'sound-5.5-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: seq: Fix racy access for queue timer in proc read
ALSA: usb-audio: fix sync-ep altsetting sanity check
ASoC: msm8916-wcd-digital: Reset RX interpolation path after use
ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1
ASoC: cros_ec_codec: Make the device acpi compatible
ASoC: sti: fix possible sleep-in-atomic
ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1
ASoC: hdac_hda: Fix error in driver removal after failed probe
ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers
ASoC: SOF: Intel: lower print level to dbg if we will reinit DSP
ALSA: dice: fix fallback from protocol extension into limited functionality
ALSA: firewire-tascam: fix corruption due to spin lock without restoration in SoftIRQ context
ALSA: hda: Rename back to dmic_detect option
ASoC: stm32: dfsdm: fix 16 bits record
ASoC: stm32: sai: fix possible circular locking
ASoC: Fix NULL dereference at freeing
ASoC: Intel: bytcht_es8316: Fix Irbis NB41 netbook quirk
ASoC: rt5640: Fix NULL dereference on module unload
The snprintf calls filling cht_rt5645_cpu_dai_name /
cht_rt5645_codec_aif_name always fill them with the same string
("ssp0-port" resp "rt5645-aif2") so instead of keeping these buffers
around and making cpus->dai_name / codecs->dai_name point to this,
simply update the *->dai_name pointers to directly point to a string
constant containing the desired string.
Signed-off-by: Damian van Soelen <dj.vsoelen@gmail.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200115164619.101705-5-hdegoede@redhat.com
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snprintf calls filling byt_rt56*_codec_aif_name/byt_rt56*_cpu_dai_name
always fill them with the same string ("rt56*-aif2" resp. ssp0-port").
So instead of keeping these buffers around and making codecs->dai_name /
cpus->dai_name point to them, simply update the *->dai_name pointers to
directly point to a string constant containing the desired string.
Signed-off-by: Jordy Ubink <jordyubink@hotmail.nl>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200115164619.101705-4-hdegoede@redhat.com
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snprintf calls filling byt_rt56*_codec_aif_name/byt_rt56*_cpu_dai_name
always fill them with the same string ("rt56*-aif2" resp. ssp0-port").
So instead of keeping these buffers around and making codecs->dai_name /
cpus->dai_name point to them, simply update the *->dai_name pointers to
directly point to a string constant containing the desired string.
Signed-off-by: Nariman Etemadi <narimantos@gmail.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200115164619.101705-3-hdegoede@redhat.com
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 16 and 24 bit paths in byt_rt5640_codec_fixup are mostly identical,
introduce a local bits variable to address the only difference and move
the common bits out of the if ... else ... .
Signed-off-by: Erik Bussing <eabbussing@outlook.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200115164619.101705-2-hdegoede@redhat.com
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PM8916 has three TX inputs that each have an (optional) internal
RBIAS resistor. MIC BIAS Internal1/2 (for TX1/2) are already supported.
TX3 does not have its own MIC BIAS supply, instead it is also supplied
from MIC_BIAS1.
Now that we have simplified the MIC BIAS Internal* implementation
we can easily add support for it:
Add a MIC BIAS Internal3 supply that enables the internal RBIAS
resistor on TX3, and make sure to also enable the MIC_BIAS1 supply.
Tested-by: Nikita Travkin <nikitos.tr@gmail.com> # longcheer-l8150
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200114181229.42302-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, MIC BIAS Internal* and MIC BIAS External* both reference
the same register, and have a part of their initialization sequence
duplicated.
For example, the sequence for enabling MIC BIAS Internal1 is:
I1. Enable MIC_BIAS1 supply (MICB_EN bit in CDC_A_MICB_1_EN)
I2. Enable internal RBIAS (TX1_INT_RBIAS_EN bit in CDC_A_MICB_1_INT_RBIAS)
The sequence for enabling MIC BIAS External1 is:
E1. Enable MIC_BIAS1 supply (MICB_EN bit in CDC_A_MICB_1_EN)
(E2. Ideally, make sure internal RBIAS is disabled. However, this should
not happen in practice because DAPM will disable unused supplies...)
Right now we have:
SND_SOC_DAPM_SUPPLY("MIC BIAS Internal1", CDC_A_MICB_1_EN, 7, 0, ...) // I1
SND_SOC_DAPM_SUPPLY("MIC BIAS External1", CDC_A_MICB_1_EN, 7, 0, ...) // E1
and I2 is done in the PM event handler (pm8916_wcd_analog_enable_micbias_int1).
We can simplify this by defining a common DAPM supply for I1/E1 ("MIC_BIAS1"),
and one DAPM supply for I2 ("MIC BIAS Internal1"). Additional DAPM routes
ensure that we also enable the MIC_BIAS1 supply for the internal and external
pull up resistor.
Another advantage of this is that we now disable the internal RBIAS
when it is not needed. This makes it much easier to add support for
MIC BIAS Internal3 as a next step.
Tested-by: Nikita Travkin <nikitos.tr@gmail.com> # longcheer-l8150
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200114181229.42302-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
This is mostly driver specific fixes, plus an error handling fix
in the core. There is a rather large diffstat for the stm32 SAI
driver, this is a very large but mostly mechanical update which
wraps every register access in the driver to allow a fix to the
locking which avoids circular locks, the active change is much
smaller and more reasonably sized.
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Merge tag 'asoc-fix-v5.5-rc6' into asoc-5.6
ASoC: Fixes for v5.5
This is mostly driver specific fixes, plus an error handling fix
in the core. There is a rather large diffstat for the stm32 SAI
driver, this is a very large but mostly mechanical update which
wraps every register access in the driver to allow a fix to the
locking which avoids circular locks, the active change is much
smaller and more reasonably sized.
In case of error, the function devm_regmap_init() returns ERR_PTR() and
never returns NULL. The NULL test in the return value check should be
replaced with IS_ERR().
Fixes: d1ede0641b ("ASoC: rt715: add RT715 codec driver")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Link: https://lore.kernel.org/r/20200117024149.75515-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of error, the function devm_regmap_init() returns ERR_PTR() and
never returns NULL. The NULL test in the return value check should be
replaced with IS_ERR().
Fixes: 320b8b0d13 ("ASoC: rt711: add rt711 codec driver")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Link: https://lore.kernel.org/r/20200115143034.94492-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of error, the function devm_regmap_init() returns ERR_PTR() and
never returns NULL. The NULL test in the return value check should be
replaced with IS_ERR().
Fixes: 7d2a5f9ae4 ("ASoC: rt700: add rt700 codec driver")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Link: https://lore.kernel.org/r/20200115143027.94364-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The QFN package is a new one.
There is a different initial setting to the chip of QFN and WLCSP package.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200116091854.18095-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Should the write to MADERA_OUTPUT_ENABLES_1 fail and out_clamp[0] not be
set an additional error message will be printed. Clear the ret variable
to avoid this.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200114161841.451-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With fourth pin added for iDisp for skl_dai, update SOF_SKL_DAI_NUM to
account for the change. Without this, dais from the bottom of the list
are skipped. In current state that's the case for 'Alt Analog CPU DAI'.
Fixes: ac42b142cd76 ("ASoC: SOF: Intel: hda: Add iDisp4 DAI")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113114054.9716-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Correctly link both channels on the DAC if an output muxed between a
stereo and mono output. Without this one channel of the DAC may be
erroneously powered down whilst in mono mode.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200114161841.451-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The recently added API that exposes firmware mixer controls to the
kernel is missing cache handling and all writes bypass the cache, this
obviously causes the cache to get out of sync with the hardware. Factor
out the cache handling into two new helper functions and call those from
both the normal ALSA control handlers and the new kernel API.
Fixes: eb65ccdb08 ("ASoC: wm_adsp: Expose mixer control API")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200114161841.451-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Both snd_pcm_hw_constraints_init() and _complete() functions are
called only from pcm_native.c, hence they can be static for further
optimization.
Link: https://lore.kernel.org/r/20200116162825.24792-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD HD-audio codec driver has a few code lines invoking
snd_get_num_conns() and using its return value as the array index
without checking. This is basically safe in all those places; at the
second and later calls snd_get_num_conns() returns the value cached
from the first invocation, hence the value is always consistent.
However, it looks a bit confusing as if a lack of the proper check.
This patch introduces a new field num_smux_conns in ad198x_spec for
simplifying the code. Now we store and refer to the value more
locally without invoking the extra function at each time.
Reported-by: Colin King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200115100035.22511-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is mostly driver specific fixes, plus an error handling fix
in the core. There is a rather large diffstat for the stm32 SAI
driver, this is a very large but mostly mechanical update which
wraps every register access in the driver to allow a fix to the
locking which avoids circular locks, the active change is much
smaller and more reasonably sized.
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Merge tag 'asoc-fix-v5.5-rc6' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.5
This is mostly driver specific fixes, plus an error handling fix
in the core. There is a rather large diffstat for the stm32 SAI
driver, this is a very large but mostly mechanical update which
wraps every register access in the driver to allow a fix to the
locking which avoids circular locks, the active change is much
smaller and more reasonably sized.
This fixes crackling sound during playback.
Further note: MOTU is known for reusing Product IDs for different
devices or different generations of the device (e.g. MicroBook
I/II/IIc shares a single Product ID). This patch was only tested with
M4 audio interface, but the same Product ID is also used by M2. Hope
it will work for M2 as well.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_info_timer_read() reads the information of the timer assigned
for each queue, but it's done in a racy way which may lead to UAF as
spotted by syzkaller.
This patch applies the missing q->timer_mutex lock while accessing the
timer object as well as a slight code change to adapt the standard
coding style.
Reported-by: syzbot+2b2ef983f973e5c40943@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200115203733.26530-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support multiple endpoints on SGTL5000 codec port when used in
of_graph context.
This patch allows to share the codec port between two CPU DAIs.
Example:
Custom STM32MP157C board uses SGTL5000 audio codec. This codec is
connected to two serial audio interfaces, which are configured
either as rx or tx.
From AsoC point of view the topolgy is the following:
// 2 CPU DAIs (SAI2A/B), 1 Codec (SGTL5000)
Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0
Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0
In the DT two endpoints have to be associated to the codec port:
sgtl5000_port: port {
sgtl5000_tx_endpoint: endpoint@0 {
remote-endpoint = <&sai2a_endpoint>;
};
sgtl5000_rx_endpoint: endpoint@1 {
remote-endpoint = <&sai2b_endpoint>;
};
};
However, when the audio graph card parses the codec nodes, it expects
to find DAI interface indexes matching the endpoints indexes.
The current patch forces the use of DAI id 0 for both endpoints,
which allows to share the codec DAI between the two CPU DAIs
for playback and capture streams respectively.
Signed-off-by: Marek Vasut <marex@denx.de>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191219213219.366073-1-marex@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
There are two asrc module in imx8qm & imx8qxp, each module has
different clock configuration, and the DMA type is EDMA.
So in this patch, we define the new clocks, refine the clock map,
and include struct fsl_asrc_soc_data for different soc usage.
The EDMA channel is fixed with each dma request, one dma request
corresponding to one dma channel. So we need to request dma
channel with dma request of asrc module.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/f33dfe3157b5ab200e09ccbf9ab73d31fac6664b.1575452454.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Comparing the voltage of VDDA and VDDIO to determine whether or not to
enable VDDC manual override is insufficient. This is a problem in case
the VDDA is supplied from different regulator than VDDIO, while both
report the same voltage to the regulator framework. In that case where
VDDA and VDDIO is supplied by different regulators, the VDDC manual
override must not be applied.
Fixes: b6319b061b ("ASoC: sgtl5000: Fix charge pump source assignment")
Signed-off-by: Marek Vasut <marex@denx.de>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Igor Opaniuk <igor.opaniuk@toradex.com>
Cc: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Link: https://lore.kernel.org/r/20191220164450.1395038-2-marex@denx.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Add jz4770-codec driver to support the internal CODEC found in the
JZ4770 SoC from Ingenic.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Maarten ter Huurne <maarten@treewalker.org>
Link: https://lore.kernel.org/r/20191224002708.1207884-2-paul@crapouillou.net
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_fwnode_get_index_gpiod_from_child() is going away as the name is
too unwieldy, let's switch to using the new devm_fwnode_gpiod_get().
Signed-off-by: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Link: https://lore.kernel.org/r/20200103011754.GA260926@dtor-ws
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix GCC warning with W=1
sound/soc/intel//boards/bytcr_rt5651.c:659:40: warning:
‘byt_rt5651_dai_params’ defined but not used
[-Wunused-const-variable=]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-19-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix GCC warning with W=1
sound/soc/intel//boards/bytcr_rt5640.c:936:40: warning:
‘byt_rt5640_dai_params’ defined but not used
[-Wunused-const-variable=]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
fix GCC warning with W=1
sound/soc/intel//boards/bytcht_es8316.c:237:40: warning:
‘byt_cht_es8316_dai_params’ defined but not used
[-Wunused-const-variable=]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-17-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
[sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c:764]: (style)
Variable 'ret' is assigned a value that is never used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning
[sound/soc/intel/boards/kbl_da7219_max98927.c:179]: (style) Variable
'ret' is assigned a value that is never used.
[sound/soc/intel/boards/kbl_da7219_max98927.c:1098]: (style) Variable
'i' is assigned a value that is never used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix cppcheck warning:
[sound/soc/intel/boards/kbl_da7219_max98927.c:340] ->
[sound/soc/intel/boards/kbl_da7219_max98927.c:348]: (style) Variable
'ret' is reassigned a value before the old one has been used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix GCC warning with W=1
sound/soc/intel/boards/glk_rt5682_max98357a.c:256:48: warning:
‘constraints_channels’ defined but not used [-Wunused-const-variable=]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113210428.27457-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix GCC warning with W=1, previous cleanup did not remove unnecessary
variable.
sound/soc/sof/intel/hda-dai.c: In function ‘hda_link_pcm_prepare’:
sound/soc/sof/intel/hda-dai.c:265:31: warning: variable ‘hda_stream’
set but not used [-Wunused-but-set-variable]
265 | struct sof_intel_hda_stream *hda_stream;
| ^~~~~~~~~~
Fixes: a3ebccb52e ("ASoC: SOF: Intel: hda: reset link DMA state in prepare")
Cc: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113205620.27285-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixup this error
CC sound/soc/codecs/rt715-sdw.o
linux/sound/soc/codecs/rt715-sdw.c: In function 'rt715_dev_resume':
linux/sound/soc/codecs/rt715-sdw.c:568:28: error: implicit declaration\
of function 'to_sdw_slave_device'; did you mean 'sdw_slave_modalias'?\
[-Werror=implicit-function-declaration]
struct sdw_slave *slave = to_sdw_slave_device(dev);
^~~~~~~~~~~~~~~~~~~
sdw_slave_modalias
linux/sound/soc/codecs/rt715-sdw.c:568:28: warning: initialization of\
'struct sdw_slave *' from 'int' makes pointer from integer without a\
cast [-Wint-conversion]
cc1: some warnings being treated as errors
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h80yhm9p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The altsetting sanity check in set_sync_ep_implicit_fb_quirk() was
checking for there to be at least one altsetting but then went on to
access the second one, which may not exist.
This could lead to random slab data being used to initialise the sync
endpoint in snd_usb_add_endpoint().
Fixes: c75a8a7ae5 ("ALSA: snd-usb: add support for implicit feedback")
Fixes: ca10a7ebdf ("ALSA: usb-audio: FT C400 sync playback EP to capture EP")
Fixes: 5e35dc0338 ("ALSA: usb-audio: add implicit fb quirk for Behringer UFX1204")
Fixes: 17f08b0d9a ("ALSA: usb-audio: add implicit fb quirk for Axe-Fx II")
Fixes: 103e962564 ("ALSA: usb-audio: simplify set_sync_ep_implicit_fb_quirk")
Cc: stable <stable@vger.kernel.org> # 3.5
Signed-off-by: Johan Hovold <johan@kernel.org>
Link: https://lore.kernel.org/r/20200114083953.1106-1-johan@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
make W=1 reports the following warnings, fix as suggested
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_non_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:824:3: warning: suggest braces around empty
body in an ‘if’ statement [-Wempty-body]
824 | ;
| ^
sound/pci/hda/patch_hdmi.c:826:3: warning: suggest braces around empty
body in an ‘if’ statement [-Wempty-body]
826 | ;
| ^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113211405.28070-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make W=1 throws a lot of warnings, with multiple misalignments between
function params and their descriptions.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200113205638.27338-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the initial amplifier driver for rt1308-sdw.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20200110014606.17333-1-shumingf@realtek.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If CONFIG_SND_ATMEL_SOC_DMA=m, build error:
sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio':
(.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register'
Function atmel_pcm_dma_platform_register is defined under
CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in
CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Chen Zhou <chenzhou10@huawei.com>
Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason, attempting to route audio through QDSP6 on MSM8916
causes the RX interpolation path to get "stuck" after playing audio
a few times. In this situation, the analog codec part is still working,
but the RX path in the digital codec stops working, so you only hear
the analog parts powering up. After a reboot everything works again.
So far I was not able to reproduce the problem when using lpass-cpu.
The downstream kernel driver avoids this by resetting the RX
interpolation path after use. In mainline we do something similar
for the TX decimator (LPASS_CDC_CLK_TX_RESET_B1_CTL), but the
interpolator reset (LPASS_CDC_CLK_RX_RESET_CTL) got lost when the
msm8916-wcd driver was split into analog and digital.
Fix this problem by adding the reset to
msm8916_wcd_digital_enable_interpolator().
Fixes: 150db8c5af ("ASoC: codecs: Add msm8916-wcd digital codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200105102753.83108-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
MIC BIAS Internal1 is broken at the moment because we always
enable the internal rbias resistor to the TX2 line (connected to
the headset microphone), rather than enabling the resistor connected
to TX1.
Move the RBIAS code to pm8916_wcd_analog_enable_micbias_int1/2()
to fix this.
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200111164006.43074-3-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
MIC BIAS External1 sets pm8916_wcd_analog_enable_micbias_ext1()
as event handler, which ends up in pm8916_wcd_analog_enable_micbias_ext().
But pm8916_wcd_analog_enable_micbias_ext() only handles the POST_PMU
event, which is not specified in the event flags for MIC BIAS External1.
This means that the code in the event handler is never actually run.
Set SND_SOC_DAPM_POST_PMU as the only event for the handler to fix this.
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200111164006.43074-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
In case system has multiple HDA codecs, and codec probe fails for
at least one but not all codecs, driver will end up cancelling
a non-initialized timer context upon driver removal.
Call trace of typical case:
[ 60.593646] WARNING: CPU: 1 PID: 1147 at kernel/workqueue.c:3032
__flush_work+0x18b/0x1a0
[...]
[ 60.593670] __cancel_work_timer+0x11f/0x1a0
[ 60.593673] hdac_hda_dev_remove+0x25/0x30 [snd_soc_hdac_hda]
[ 60.593674] device_release_driver_internal+0xe0/0x1c0
[ 60.593675] bus_remove_device+0xd6/0x140
[ 60.593677] device_del+0x175/0x3e0
[ 60.593679] ? widget_tree_free.isra.7+0x90/0xb0 [snd_hda_core]
[ 60.593680] snd_hdac_device_unregister+0x34/0x50 [snd_hda_core]
[ 60.593682] snd_hdac_ext_bus_device_remove+0x2a/0x60 [snd_hda_ext_core]
[ 60.593684] hda_dsp_remove+0x26/0x100 [snd_sof_intel_hda_common]
[ 60.593686] snd_sof_device_remove+0x84/0xa0 [snd_sof]
[ 60.593687] sof_pci_remove+0x10/0x30 [snd_sof_pci]
[ 60.593689] pci_device_remove+0x36/0xb0
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-9-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case system has multiple HDA controllers, it can happen that
same HDA codec driver is used for codecs of multiple controllers.
In this case, SOF may fail to probe the HDA driver and SOF
initialization fails.
SOF HDA code currently relies that a call to request_module() will
also run device matching logic to attach driver to the codec instance.
However if driver for another HDA controller was already loaded and it
already loaded the HDA codec driver, this breaks current logic in SOF.
In this case the request_module() SOF does becomes a no-op and HDA
Codec driver is not attached to the codec instance sitting on the HDA
bus SOF is controlling. Typical scenario would be a system with both
external and internal GPUs, with driver of the external GPU loaded
first.
Fix this by adding similar logic as is used in legacy HDA driver
where an explicit device_attach() call is done after request_module().
Also add logic to propagate errors reported by device_attach() back
to caller. This also works in the case where drivers are not built
as modules.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We will reinit DSP in a loop when it fails to initialize the first
time, as recommended. So, it is not an error before we finally give
up. And reorder the trace to make it more readable.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
RT711 is in SoundWire mode on link0.
RT1308 is either on SSP2 or on SoundWire link1 (depending on hardware
reworks).
Signed-off-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110222530.30303-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The two configurations are with the Realtek 3-in-1 board requiring all
4 links to be enabled, or basic configuration with the on-board
RT700 using link1.
For now we only have definitions for CML. CNL and CFL are just
placeholders.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110222530.30303-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The two configurations are with the Realtek 3-in-1 board requiring all
4 links to be enabled, or basic configuration with the on-board RT700
using link0.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110222530.30303-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Any app using ALSA OSS emulation on top of SOF will fail
to error from OSS SNDCTL_DSP_SETFMT ioctl. Reported initially
as an issue with xournalpp (application using PortAudio with
an OSS backend), but applies more generally to other apps
using OSS as well.
Problem is caused by SOF PCM not supporting repeated calls
to hw_params(), without matching calls to pcm_free(). This
is however exactly what the ALSA OSS PCM code is doing when
it is handling the OSS ioctls.
The problem will lead to leaking of DSP resources and eventual
failure of DSP PCM_PARAMS IPC.
BugLink: https://github.com/thesofproject/linux/issues/1510
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The legacy driver uses dummy cpu_dai and platform, SOF requires actual
values to bind.
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing machine driver depends on SPI Master capabilities, but
the Kconfig does not model this dependency and the SPI controller
needs to be selected as well.
Without this patch the machine driver probe would fail with the
spi-RT5677AA:00 component never registered by the ACPI/LPSS subsystem.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110235751.3404-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a857e073ff ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") removed
the last use of the rtd variable but didn't remove its definition,
leading to the following warning/error for MIPS rbtx49xx_defconfig
builds:
sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_hw_params':
sound/soc/txx9/txx9aclc.c:54:30: error: unused variable 'rtd'
[-Werror=unused-variable]
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
^~~
Resolve this by removing the unused variable.
Signed-off-by: Paul Burton <paulburton@kernel.org>
Fixes: a857e073ff ("ASoC: txx9: txx9aclc: remove snd_pcm_ops")
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Link: https://lore.kernel.org/r/20200109191422.334516-1-paulburton@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
In the commit 8e85def572 ("ALSA: hda: enable regmap internal
locking"), we re-enabled the regmap lock due to the reported
regression that showed the possible concurrent accesses. It was a
temporary workaround, and there are still a few opened races even
after the revert. In this patch, we cover those still opened windows
with a proper mutex lock and disable the regmap internal lock again.
First off, the patch introduces a new snd_hdac_device.regmap_lock
mutex that is applied for each snd_hdac_regmap_*() call, including
read, write and update helpers. The mutex is applied carefully so
that it won't block the self-power-up procedure in the helper
function. Also, this assures the protection for the accesses without
regmap, too.
The snd_hdac_regmap_update_raw() is refactored to use the standard
regmap_update_bits_check() function instead of the open-code. The
non-regmap case is still open-coded but it's an easy part. The all
read and write operations are in the single mutex protection, so it's
now race-free.
In addition, a couple of new helper functions are added:
snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both
are called from HD-audio legacy driver. The former is to initialize
the given verb bits but only once when it's not initialized yet. Due
to this condition, the function invokes regcache_cache_only(), and
it's now performed inside the regmap_lock (formerly it was racy) too.
The latter function is for simply invoking regcache_sync() inside the
regmap_lock, which is called from the codec resume call path.
Along with that, the HD-audio codec driver code is slightly modified /
simplified to adapt those new functions.
And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are
rewritten with the helper macro. It's just for simplification because
the code logic is identical among all those functions.
Tested-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200109090104.26073-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add delay to make sure that audio urbs are not sent too early.
Otherwise the device hangs. Windows driver makes ~2s delay, so use
about the same time delay value.
snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which
is an overkill. Thus a quirk that is called only once is implemented.
Also send two vendor-specific control messages before and after
the delay. This behaviour is blindly copied from the Windows driver.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200112102358.18085-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA dice driver expects devices to multiplex MIDI messages into first
port of isochronous communication. Actually devices perform for it.
However, check of stream format is invalid for second port of isochronous
communication. As a result, when the device supports two ports for
isochronous communication and the stream format is hard-coded, ALSA
dice driver fails to start packet streaming.
This commit loosens stream format check for MIDI conformant data channel.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113084630.14305-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At failure of attempt to detect protocol extension, ALSA dice driver
should be fallback to limited functionality. However it's not.
This commit fixes it.
Cc: <stable@vger.kernel.org> # v4.18+
Fixes: 58579c056c ("ALSA: dice: use extended protocol to detect available stream formats")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113084630.14305-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stanton SCS.1d uses Oxford Semiconductor FW 971 ASIC (FW971) for
communication. Although the unit is bound to ALSA oxfw driver, the instance
of sound card can not be added due to its quirk of plug information. This
bug was added when snd-scs1x is merged into snd-oxfw at commit
9e2004f9ce ("ALSA: oxfw: obsolete scs1x module").
This commit fixes the driver for the quirk. In cases that the unit returns
NOT IMPLEMENTED for some AV/C commands, the sound card is added without any
PCM/MIDI interfaces for packet streaming. For SCS.1d, model dependent
operation adds MIDI interface and applications can use it to operate
according to HSS1394 protocol from reverse-engineering work by Sean M.
Pappalardo.
Plug Control Register (PCR) has information that the unit has a pair of
plugs for isochronous communication:
(oMPR)
$ ./firewire-request /dev/fw1 read 0xfffff0000900
result: 80ff0001
(iMPR)
$ ./firewire-request /dev/fw1 read 0xfffff0000980
result: 80ff0001
AV/C PLUG INFO also returns information that the unit has a pair of
plugs for isochronous communication.
(AV/C PLUG INFO command)
$ ./firewire-request /dev/fw1 fcp 0x01ff0200ffffffff
response: 000: 0c ff 02 00 01 01 02 02
However, AV/C PLUG SIGNAL INFO command is rejected for both plugs.
(AV/C OUTPUT PLUG SIGNAL INFO command)
$ ./firewire-request /dev/fw1 fcp 0x01ff1800ffffffff
response: 000: 0a ff 18 00 ff ff ff ff
(AV/C INPUT PLUG SIGNAL INFO command)
$ ./firewire-request /dev/fw1 fcp 0x01ff1900ffffffff
response: 000: 0a ff 19 00 ff ff ff ff
Furthermore, AV/C EXTENDED STREAM FORMAT INFO is not implemented.
(AV/C EXTENDED STREAM FORMAT INFO list subfunction for input plug)
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc000000000ffff00ff
response: 000: 08 ff bf c0 00 00 00 00 ff ff 00 ff
(AV/C EXTENDED STREAM FORMAT INFO list subfunction for output plug)
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffff00ff
response: 000: 08 ff bf c0 01 00 00 00 ff ff 00 ff
(AV/C EXTENDED STREAM FORMAT INFO single subfunction for input plug)
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc100000000ffffffff
response: 000: 08 ff bf c1 00 00 00 00 ff ff ff ff
(AV/C EXTENDED STREAM FORMAT INFO single subfunction for output plug)
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc101000000ffffffff
response: 000: 08 ff bf c1 01 00 00 00 ff ff ff ff
Reference: https://mailman.alsa-project.org/pipermail/alsa-devel/2012-May/052264.html
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113073418.24622-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stanton SCS.1d doesn't support packet streaming even if it has plugs for
isochronous communication.
This commit is a preparation for this case. The 'has_input' member is
added to specific structure, and MIDI/PCM interfaces are not added when
the member is false.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113073418.24622-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When AV/C command returns 'NOT IMPLEMENTED' status in its response, ALSA
oxfw driver uses ENOSYS as error code. However, it's expected just to be
used for missing system call number.
This commit replaces it with ENXIO.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113073418.24622-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA firewire-tascam driver can bring corruption due to spin lock without
restoration of IRQ flag in SoftIRQ context. This commit fixes the bug.
Cc: Scott Bahling <sbahling@suse.com>
Cc: <stable@vger.kernel.org> # v4.21
Fixes: d716742243 ("ALSA: firewire-tascam: queue events for change of control surface")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200113085719.26788-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
GCC reports the following warning with W=1
sound/usb/mixer_quirks.c: In function ‘snd_microii_controls_create’:
sound/usb/mixer_quirks.c:1694:2: warning: ‘static’ is not at beginning
of declaration [-Wold-style-declaration]
1694 | const static usb_mixer_elem_resume_func_t resume_funcs[] = {
| ^~~~~
Move static to the beginning of declaration
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200111214736.3002-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
GCC reports the following warning with W=1
sound/pci/hda/patch_realtek.c: In function ‘alc269_suspend’:
sound/pci/hda/patch_realtek.c:3616:29: warning: suggest braces around
empty body in an ‘if’ statement [-Wempty-body]
3616 | alc5505_dsp_suspend(codec);
| ^
sound/pci/hda/patch_realtek.c: In function ‘alc269_resume’:
sound/pci/hda/patch_realtek.c:3651:28: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
3651 | alc5505_dsp_resume(codec);
| ^
This is a classic macro problem and can indeed lead to bad program
flows.
Fix by using the usual "do { } while (0)" pattern
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200111214736.3002-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
GCC reports a warning with W=1:
sound/core/timer.c: In function ‘snd_timer_user_read’:
sound/core/timer.c:2219:19: warning: initialized field overwritten
[-Woverride-init]
2219 | .tstamp_sec = tread->tstamp_nsec,
| ^~~~~
sound/core/timer.c:2219:19: note: (near initialization for
‘(anonymous).tstamp_sec’)
Assigning nsec values to sec fields is problematic in general, even
more so when the initial goal was to survive the 2030 timer
armageddon.
Fix by using the proper field in the initialization
Cc: Baolin Wang <baolin.wang@linaro.org>
Cc: Arnd Bergmann <arnd@arndb.de>
Fixes: 07094ae6f9 ("ALSA: Avoid using timespec for struct snd_timer_tread")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20200111203325.20498-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got quite a few bug reports showing the SOF driver being loaded
unintentionally recently, and the reason seems to be that users didn't
know the module option change: with the recent kernel, a new option
dsp_driver=1 has to be passed to a new module snd-intel-dspcfg
instead of snd_hda_intel.dmic_detect=0 option.
That is, actually there are two tricky things here:
- We changed the whole detection in another module and another
option semantics.
- The existing option for skipping the DSP probe was also renamed.
For avoiding the confusion and giving user more hint, this patch
reverts the renamed option dsp_driver back to dmic_detect for
snd-hda-intel module, and show the warning about the module option
change when the non-default value is passed.
Fixes: 82d9d54a6c ("ALSA: hda: add Intel DSP configuration / probe code")
Link: https://lore.kernel.org/r/20200109082000.26729-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few piled ASoC fixes and usual HD-audio and USB-audio fixups.
Some of them are for ASoC core, but rather about error-handling.
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Merge tag 'sound-5.5-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few piled ASoC fixes and usual HD-audio and USB-audio fixups. Some
of them are for ASoC core error-handling"
* tag 'sound-5.5-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda: enable regmap internal locking
ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen
ALSA: hda/realtek - Set EAPD control to default for ALC222
ALSA: usb-audio: Apply the sample rate quirk for Bose Companion 5
ALSA: hda/realtek - Add new codec supported for ALCS1200A
ASoC: Intel: boards: Fix compile-testing RT1011/RT5682
ASoC: SOF: imx8: Fix dsp_box offset
ASoC: topology: Prevent use-after-free in snd_soc_get_pcm_runtime()
ASoC: fsl_audmix: add missed pm_runtime_disable
ASoC: stm32: spdifrx: fix input pin state management
ASoC: stm32: spdifrx: fix race condition in irq handler
ASoC: stm32: spdifrx: fix inconsistent lock state
ASoC: core: Fix access to uninitialized list heads
ASoC: soc-core: Set dpcm_playback / dpcm_capture
ASoC: SOF: imx8: fix memory allocation failure check on priv->pd_dev
ASoC: SOF: Intel: hda: hda-dai: fix oops on hda_link .hw_free
ASoC: SOF: fix fault at driver unload after failed probe
dpcm_fe_dai_shutdown() / soc_compr_free_fe() didn't care pmdown_time.
We already have snd_soc_dapm_stream_stop() for it.
Let's use common method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhewrq9j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When we stop stream, if it was Playback, we might need to care
about power down time. In such case, we need to use delayed work.
We have same implementation for it at soc-pcm.c and soc-compress.c,
but we don't want to have duplicate code.
This patch adds snd_soc_dapm_stream_stop(), and share same code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/871rs8t4uw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to setup rtd->close_delayed_work_func.
It will be set at snd_soc_dai_compress_new() or soc_new_pcm().
But these setups close_delayed_work() which is same name /
same implemantaion, but different local code.
To reduce duplicate code, this patch moves it as
snd_soc_close_delayed_work() and share same code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/8736cot4v2.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC need to care pinctrl_pm_select_xxx().
It is called at soc-core and soc-pcm.
soc-pcm is controlling it for activate DAI.
soc-core is controlling it for whole system
(= suspend/resume/probe/poweroff).
If we focus to soc-core side, it need to care about BIAS level.
Then, snd_soc_suspend() only is controlling it by Component base (a).
Other functions are DAI base (b).
(a) pinctrl_pm_select_xxx(component->dev, xxx);
(b) pinctrl_pm_select_xxx(dai->dev, xxx);
Because of these unbalance, the code is confusable.
Here, dai->dev and component->dev are same pointer.
Thus, we can replace it component base.
One note here is that it cared DAI (= CPU/Codec) pin before this patch,
after this patch, it cares Component (= CPU/Codec/Platform) pin.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/874kx4t4v6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_suspend() are doing below for pinctrl_pm_select_sleep_state()
int snd_soc_suspend(struct device *dev)
{
...
for_each_card_components(card, component) {
...
(1) pinctrl_pm_select_sleep_state(component->dev);
}
for_each_card_rtds(card, rtd) {
...
(2) pinctrl_pm_select_sleep_state(cpu_dai->dev);
}
}
(1) is called for all component (CPU/Codec/Platform), and
(2) is called for CPU DAIs.
Here, component->dev is same as dai->dev.
This means, it is called in duplicate on CPU case.
This patch removes (2).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/875zhkt4vc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Card dai_link has .ignore_suspend, and ALSA SoC cares it when suspend.
For example, like this
for_each_card_rtds(card, rtd) {
if (rtd->dai_link->ignore_suspend)
continue;
...
}
But in snd_soc_suspend(), it doesn't care about
it when suspending Component. This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e20t4vh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't have snd_soc_rtdcom_list anymore.
Let's rename snd_soc_rtdcom_add() to more understandable
snd_soc_rtd_add_component()
Reported-by: Sridharan, Ranjani <ranjani.sridharan@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878smgt4vp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the initial codec driver for rt700.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200110014552.17252-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ALSA SoC is using struct snd_soc_rtdcom_list to
connecting component to rtd by using list_head.
struct snd_soc_rtdcom_list {
struct snd_soc_component *component;
struct list_head list; /* rtd::component_list */
};
struct snd_soc_pcm_runtime {
...
struct list_head component_list; /* list of connected components */
...
};
The CPU/Codec/Platform component which will be connected to rtd (a)
is indicated via dai_link at snd_soc_add_pcm_runtime()
int snd_soc_add_pcm_runtime(...)
{
...
/* Find CPU from registered CPUs */
rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus);
...
(a) snd_soc_rtdcom_add(rtd, rtd->cpu_dai->component);
...
/* Find CODEC from registered CODECs */
(b) for_each_link_codecs(dai_link, i, codec) {
rtd->codec_dais[i] = snd_soc_find_dai(codec);
...
(a) snd_soc_rtdcom_add(rtd, rtd->codec_dais[i]->component);
}
...
/* Find PLATFORM from registered PLATFORMs */
(b) for_each_link_platforms(dai_link, i, platform) {
for_each_component(component) {
...
(a) snd_soc_rtdcom_add(rtd, component);
}
}
}
It shows, it is possible to know how many components will be
connected to rtd by using
dai_link->num_cpus
dai_link->num_codecs
dai_link->num_platforms
If so, we can use component pointer array instead of list_head,
in such case, code can be more simple.
This patch removes struct snd_soc_rtdcom_list that is only
of temporary value, and convert to pointer array.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-By: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/87a76wt4wm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In stm32_afsdm_pcm_cb function, the transfer size is provided in bytes.
However, samples are copied as 16 bits words from iio buffer.
Divide by two the transfer size, to copy the right number of samples.
Fixes: 1e7f6e1c69 ("ASoC: stm32: dfsdm: add 16 bits audio record support")
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20200110131131.3191-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede