The calls for capture_hook were missing in dyn_adc_capture_pcm_prepare
and cleanup callbacks. Luckily there are no users of the capture
hooks with dyn-adc PCM, so far, thus this doesn't change the behavior
of existing devices, but it's a fix for a future usage.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is Dell usb dock audio workaround.
It was fixed the master volume keep lower.
[Some background: the patch essentially skips the controls of a couple
of FU volumes. Although the firmware exposes the dB and the value
information via the usb descriptor, changing the values (we set the
min volume as default) screws up the device. Although this has been
fixed in the newer firmware, the devices are shipped with the old
firmware, thus we need the workaround in the driver side. -- tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FW expects sampling rate rounded up to next higher integer value
when calculating ibs/obs. For example for 44.1k, it should be
rounded up to 45 to calculate ibs/obs.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Here are the changes to enable reader tdm mode:
- When TDM_ENABLE is set to 1, the i2s format should be automatically
configured. Unfortunately this is not the case (HW bug). Then, we shall
force DATA_SIZE setting.
- Compute the transfer size for tdm mode: transfer size = user frame size
- Manage tdm slots configuration given in DT.
- Refine the hw param (channels & format) according to tdm slot config.
Signed-off-by: Moise Gergaud <moise.gergaud@st.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
here are the changes to enable player tdm mode:
- When TDM_ENABLE is set to 1, the i2s format should be automatically
configured. Unfortunately this is not the case (HW bug). Then, we shall
force DATA_SIZE setting.
- Compute the transfer size for tdm mode: transfer size = user frame size
- Manage tdm slots configuration given in DT.
- Don't use mclk-fs when unip in tdm mode; use tdm slot config to compute
frame size and to set mclk rate.
- Refine the hw param (channels & format) according to tdm slot config.
Signed-off-by: Moise Gergaud <moise.gergaud@st.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Lenovo Thinkpad T460s requires the alc_fixup_tpt440_dock as well in
order to get working sound output on the docking stations headphone jack.
Patch tested on a Thinkpad T460s (20F9CT01WW) using a ThinkPad Ultradock
on kernel 4.4.6.
Signed-off-by: Sven Eckelmann <sven@narfation.org>
Tested-by: Simon Wunderlich <sw@simonwunderlich.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'size' member of a struct firmware is passed to snd_printk with a
respective format string using the %d identifier. The 'size' member is
of type size_t, but format identifier %d indicates a signed int data
type. This patch replaces the %d format identifier with the correct %zu
format identifier for size_t data types.
Signed-off-by: William Breathitt Gray <vilhelm.gray@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CS47L24 and WM1831 codecs only use the OUT4L widget so we can
skip creation of the OUT4R widget.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
miniDSP USBStreamer UAC2 devices send clock validity changes with the
control field set to zero. The current interrupt handler ignores all
packets if the control field does not match the mixer element's, but
it really should only do that in case that field is needed to
distinguish multiple elements with the same ID.
This patch implements a logic that lets notifications packets pass
if the element ID is unique for a given device.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 specifies clock sources that optionally have validity controls.
This patch exposes them as mixer controls, so they can be read (and
at least in theory even be written) by userspace applications in order
to make clock selection policy decisions.
This implementation does nothing if the device is not UAC2 compliant,
or if the clock source does not define said validity control bits.
Tested with a miniDSP USBStreamer (0x2752/0x0016).
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ct_timer_ops structures are never modified, so declare them as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
acpi_dev_present() was originally named after pci_dev_present()
to signify the similarity of the two functions.
However Rafael J. Wysocki pointed out that the exported function
acpi_dev_present() is easily confused with the non-exported
acpi_device_is_present(). Additionally in ACPI parlance the term
"present" usually refers to the "device is present" bit returned
by the _STA control method, yet acpi_dev_present() merely checks
presence in the namespace. It does not invoke _STA at all, let
alone check the "device is present" bit.
As suggested by Rafael, rename the function to acpi_dev_found()
and adjust all existing call sites.
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
This patch makes Designware I2S driver use the fifo
depth value to program the fifo configuration register
instead of using hardcoded values.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Treat 32 bit sample width as if it was 24 bits when generating IEC958
channel status bits. On some platforms 24 sample width is problematic
and to get full 24 bit precision a 32 bit format, using only the 24
most significant bits, may have to be used.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add IEC958 channel status helper that gets the audio properties from
snd_pcm_hw_params instead of snd_pcm_runtime. This is needed to
produce the channel status bits already in audio stream configuration
phase.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
If we encounter an error on the DSP side whilst user-space is
waiting on the poll we should call snd_compr_fragment_elapsed,
although data is not actually available we want to wake
user-space such that the error can be propagated out
quickly. Additionally some versions of the DSP firmware are
not super consistent about actually generating an IRQ if they
encounter an error, as such we will check the DSP error status
every time we run out of available data as well, to ensure we
catch it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Factor out the reading of the DSP error flag into its own function to
support further improvements to the code.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the original code we ended the loop with tries set to -1 instead of
zero.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Plantronics BT300 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1". This patch adds the USB
ID of the BT300 to quirks.c and avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
intel8x0 driver has the inside_vm check for skipping a buggy hardware
workaround in the PCM pointer callback in the commit [228cf79376f1:
ALSA: intel8x0: Improve performance in virtual environment]. This was
originally applied to all devices on known VMs, but the code was
switched to use the PCI ID matching for applying to only known
devices (KVM and Parallels), in order to avoid applying wrongly to
VT-d and other such cases, in the commit [7fb4f392bd27: ALSA:
intel8x0: improve virtual environment detection].
Meanwhile, the original VM check was kept even after switching to the
PCI ID matching. It was partly because we weren't 100% sure whether
we had covered all well, and partly because this would help
identifying the issue once when a user of another VM hit the same
problem or a regression. Currently the VM check is used only for
showing the kernel message that the VM-optimization isn't applied, and
the VM check itself doesn't change the actual driver behavior at all.
Despite the relatively safe driver behavior, the code caught attention
of developers badly and brought many confusion / misunderstanding.
Since we've got neither regression nor enhancement report for other
VMs for five years long, it's likely safe to drop this superfluous VM
check now.
The module option is still kept, so if a user still needs to adjust,
it can be applied as was.
Acked-by: Konstantin Ozerkov <kozerkov@parallels.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need to unmask all interrupts at I2S start. This
can cause performance issues in slower platforms.
Unmask only the interrupts for the used channels.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media fixes from Mauro Carvalho Chehab:
"Some bug fixes on au0828 and snd-usb-audio:
- the au0828+snd-usb-audio MC patch broke several things and produced
some race conditions. Better to revert the patches, and re-work on
them for a next version
- fix a regression at tuner disable links logic
- properly handle dev_state as a bitmask"
* tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] Revert "[media] media: au0828 change to use Managed Media Controller API"
[media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
[media] au0828: Fix dev_state handling
[media] au0828: fix au0828_v4l2_close() dev_state race condition
[media] media: au0828 fix to clear enable/disable/change source handlers
[media] v4l2-mc: cleanup a warning
[media] au0828: disable tuner links and cache tuner/decoder
The existing TLV callback implementation copies all of the
cea_channel_speaker_allocation map table to the TLV container
irrespective of what is reported by sink. This is of little use
to the userspace application.
With this patch, it parses the spk_alloc block as queried from
the ELD, and copies only the corresponding mapping channel
allocation entries from the cea channel speaker allocation table.
Thus the user can parse the TLV container to identify sink's
capability and set the channel map accordingly.
It shouldn't impact the behavior in AMD chipset, as this makes
use of already parsed spk alloc block to calculate the channel
map.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Phoenix Audio TMX320 gives the similar error when the sample rate is
asked:
usb 2-1.3: 2:1: cannot get freq at ep 0x85
usb 2-1.3: 1:1: cannot get freq at ep 0x2
....
Add the corresponding USB-device ID (1de7:0014) to
snd_usb_get_sample_rate_quirk() list.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Skylake and onwards, the HD-audio controller driver needs to bind
with i915 for having the control of power well audio domain before
actually probing the codec. This leads to the load of i915 driver
from the audio driver side. But, there are systems that have no Intel
graphics but Nvidia or AMD GPU, although they still use HD-audio bus
for the onboard audio codecs. On these, loading the i915 driver is
nothing but a useless memory and CPU consumption.
A simple way to avoid it is just to look for the Intel graphics PCI
entry beforehand, and try to bind with i915 only when such an entry is
found. Currently, it assumes the PCI display class. If another class
appears, this needs to be extended (although it's very unlikely).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the system enters S3 during a playback, codec power needs to
be turned OFF during suspend and restored during resume. With
this patch the AFG node is set to D3 and codec power is turned
OFF during controller suspend call.
During resume, the codec power is left in ON state if the
playback was in progress while suspending.
Also setting power state for AFG node is optimized. With this the
loop with timeout is removed and codec_read is used instead.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use dev_pm ops feature instead of soc pm as core assumes system
is capable of direct complete. Register with complete callback
instead of resume to synchronize with Jack notification from
display driver. This ensures correct Jack notification to user
space.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of small fixes:
- A fix in ALSA timer core to avoid possible BUG() trigger
- A fix in ALSA timer core 32bit compat layer
- A few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- Fixes of USB-audio double-free at some error paths
- A fix for memory leak in DICE driver at hotunplug
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Merge tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes:
- a fix in ALSA timer core to avoid possible BUG() trigger
- a fix in ALSA timer core 32bit compat layer
- a few HD-audio quirks for ASUS and HP machines
- AMD HD-audio HDMI controller quirks
- fixes of USB-audio double-free at some error paths
- a fix for memory leak in DICE driver at hotunplug"
* tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: timer: Use mod_timer() for rearming the system timer
ALSA: hda - fix front mic problem for a HP desktop
ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
ALSA: dice: fix memory leak when unplugging
ALSA: hda - Apply fix for white noise on Asus N550JV, too
ALSA: hda - Fix white noise on Asus N750JV headphone
ALSA: hda - Asus N750JV external subwoofer fixup
ALSA: timer: fix gparams ioctl compatibility for different architectures
ALSA system timer backend stops the timer via del_timer() without sync
and leaves del_timer_sync() at the close instead. This is because of
the restriction by the design of ALSA timer: namely, the stop callback
may be called from the timer handler, and calling the sync shall lead
to a hangup. However, this also triggers a kernel BUG() when the
timer is rearmed immediately after stopping without sync:
kernel BUG at kernel/time/timer.c:966!
Call Trace:
<IRQ>
[<ffffffff8239c94e>] snd_timer_s_start+0x13e/0x1a0
[<ffffffff8239e1f4>] snd_timer_interrupt+0x504/0xec0
[<ffffffff8122fca0>] ? debug_check_no_locks_freed+0x290/0x290
[<ffffffff8239ec64>] snd_timer_s_function+0xb4/0x120
[<ffffffff81296b72>] call_timer_fn+0x162/0x520
[<ffffffff81296add>] ? call_timer_fn+0xcd/0x520
[<ffffffff8239ebb0>] ? snd_timer_interrupt+0xec0/0xec0
....
It's the place where add_timer() checks the pending timer. It's clear
that this may happen after the immediate restart without sync in our
cases.
So, the workaround here is just to use mod_timer() instead of
add_timer(). This looks like a band-aid fix, but it's a right move,
as snd_timer_interrupt() takes care of the continuous rearm of timer.
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The front mic jack (pink color) can't detect any plug or unplug. After
applying this fix, both detecting function and recording function
work well.
BugLink: https://bugs.launchpad.net/bugs/1564712
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IS_ERR_VALUE should be used only with unsigned long type,
signed types should use comparison 'ret < 0'
This patch removes such usages.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes following static checker warning, by initializing the
ret to -EINVAL, as one of the code path in lpass_platform_pcm_new()
uses this variable uninitialized.
sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new()
error: uninitialized symbol 'ret'.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
https://bugzilla.kernel.org/show_bug.cgi?id=115561
It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.
So, better to revert it and fix the core before reapplying this
change.
This reverts commit aebb2b89bff0 ("[media] sound/usb: Use Media
Controller API to share media resources")'
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
The list of TAS571x registers was incomplete.
Added the missing register definitions up to the register 0x25.
Added volatile and read-only register tables into tas5711_regmap_config
and tas5717_regmap_config.
The chip has 256 registers in total. But from address 0x29 on
(0x26 to 0x28 are reserved) the register width varies between
20, 12 and 8 bytes, which the register map cannot represent.
Signed-off-by: Petr Kulhavy <petr@barix.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.
This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.
Based on a patch by Takashi Iwai <tiwai@suse.de>
[Note for stable backports:
this patch requires the commit 902eb7fd1e4a ('ALSA: usb-audio: Minor
code cleanup in create_fixed_stream_quirk()')]
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When some digi00x units are connected sequentially, userspace
applications are involved at bus-reset state on IEEE 1394 bus. In the
state, any communications can be canceled. Therefore, sound card
registration should be delayed till the bus gets calm.
This commit achieves it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some oxfw based units tends to fail asynchronous communication when
IEEE 1394 bus is under bus-reset state. When registering sound card
instance at unit probe callback, userspace applications can be involved
to the state.
This commit postpones the registration till the bus is calm.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When some fireworks units are connected sequentially, userspace
applications are involved at bus-reset state on IEEE 1394 bus. In the
state, any communications can be canceled. Therefore, sound card
registration should be delayed till the bus gets calm.
This commit achieves it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some bebob based units tends to fail asynchronous communication when
IEEE 1394 bus is under bus-reset state. When registering sound card
instance at unit probe callback, userspace applications can be involved
to the state.
This commit postpones the registration till the bus is calm.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, ALSA dice driver postpone sound card registration after
IEEE 1394 bus is calm. This idea has advantages for the other drivers.
This commit adds a helper function for it to firewire-lib module. The
function is really for the specific purpose. Callers should initialize
delayed work structure with callback function.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, ALSA dice driver doesn't generate kernel warnings
when unplugging units before initializing stream data.
This commit moves the initialization to delayed registration of sound
card, to simplify unit probe processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
