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Move the firmware load and record parsing functionality out into
a separate function from the boot function.
Signed-off-by: Scott Ling <sl@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are uncovered cases whether the card refcount introduced by the
commit a0830dbd isn't properly increased or decreased:
- OSS PCM and mixer success paths
- When lookup function gets NULL
This patch fixes these places.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc269_toggle_power_output() was only use in ALC269VB. I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback. It turned out that the problem is that we don't
wait until all URBs are killed.
This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181
Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RayDAT reports the sync status of its inputs in consecutive bit
positions, so all we do in hdspm_s1_sync_check is to iterate over idx:
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
lock = (status & (0x1<<idx)) ? 1 : 0;
sync = (status & (0x100<<idx)) ? 1 : 0;
The index is given in kcontrol->private_value:
HDSPM_SYNC_CHECK("WC SyncCheck", 0),
HDSPM_SYNC_CHECK("AES SyncCheck", 1),
HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2),
HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3),
HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4),
HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5),
HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6),
HDSPM_SYNC_CHECK("TCO SyncCheck", 7),
HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8),
The patch corrects the indicated sync flags by passing the proper index
value to hdspm_s1_sync_check().
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the return value of cs42l52_set_fmt() when clock inversion is
not allowed and also remove the useless variable ret.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
[We had been assigning to ret but then ignoring the value we assgined
-- broonie]
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.
Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e. This confuses
the auto-parser. Fix it up in the driver by overriding these
connections.
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i). When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[]. This confuses is_empty_dac() and trims the detected DAC
in later reference.
This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission. On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used. Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Remove the boot_done counter variable and check the wm0010 state
variable instead.
Signed-off-by: Scott Ling <scott.ling@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Several bug reports suggest that the forcibly resetting IEC958 status
bits is required for AD codecs to get the SPDIF output working
properly after changing streams.
Original fix credit to Javeed Shaikh.
BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361
Reported-by: Robin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add generic ESS vendor ID to pm_whitelist. This should fix suspend on
all Maestro-2 and Maestro-2E based PCI cards.
Tested on Terratec DMX and SF64-PCE2.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correctly enable the digital microphones with the right bits in the
right coeffecient registers on Cirrus CS4206/7 codecs. It also
prevents misconfiguring ADC1/2.
This fixes the digital mic on the Macbook Pro 10,1/Retina.
Based-on-patch-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The audio chipset used in Teradici's Tera2 host cards is the same as that in
the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards.
Signed-off-by: Lars R. Damerow <lars@pixar.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Playing 24-bit format file leads to channel swap on mx28 and the reason is that
the current driver performs one write/read to/from the SAIF_DATA register to
trigger the transfer.
This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register
and thus is capable of storing the 16-bit left and right channels, but for the
S24_LE case it can only store one channel, so in order to not lose the FIFO sync
an extra read/write is needed.
Reported-by: Dan Winner <DWinner@tc-helicon.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Dan Winner <DWinner@tc-helicon.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure we select the WM1250-EV1 as the current software system
configuration demands it.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver didn't care fsi_hw_start/stop() return value,
and it causes WARNING() call if SNDRV_PCM_TRIGGER_START failed.
This patch solved this issue
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In its previous status, the first capture didn't work properly;
nothing was actually recorded from the microphone. This
behaviour was observed using a Visstrim M10 board.
In order to solve this BUG a workaround has been added that,
during the initialization process of the codec, powers on and
off the ADC.
The issue seems related to a HW BUG or some behavior that
is not documented in the datasheet.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The rate isn't restored properly after resume since it's only set up
in hw_params, and not in prepare callback. For fixing it, put the
corresponding call to resume callback as well.
Reported-and-tested-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Evalation of the WM5102 has identified a number of register values which
should be written after SYSCLK is enabled on revision A in order to
improve performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When disconnect callback is called, each component should wake up
sleepers and check card->shutdown flag for avoiding the endless sleep
blocking the proper resource release.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.
Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.
The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
mixer.c and others; the device speed is now cached in subs->speed
instead of accessing to chip->dev
The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.
The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks. They'll be covered by the
upcoming change to rwsem.
Also the mixer codes are untouched, too. These will be fixed in
another patch, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix races at PCM disconnection:
- while a PCM device is being opened or closed
- while the PCM state is being changed without lock in prepare,
hw_params, hw_free ops
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver is using fsi_set_master_clk() if it needs system clock.
But this function was called from
fsi_hw_shutdown()/fsi_dai_trigger()/fsi_resume() without a sense of unity.
Because of this, sound playback after suspend failed sometimes.
To keep consistency, fsi_master_clk() was called from
fsi_hw_start/stop() only now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many Arizona class devices contain ADSP2 cores with a standard method for
hooking them into the audio map. Define standard helpers for this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many current Wolfson devices feature DSPs based around an architecture
known as ADSP. Since there is a lot of commonality in the system
integration of these devices a common library will be used to provide
support for them.
This version provides equivalent support for ADSP1 to that currently
included in the WM2200 driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
Clean up some fallout from the OMAP header reorganisation and a minor
fix for DMIC which has no practical effect but is neater.
We should really use "fck" when asking for the functional clock and not
"dmic_fck".
This way we can ensure that multiple dmic modules can exist in the system.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also drop the includes that are no longer needed and just
cause problems for the ARM common zImage.
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
[tony@atomide.com: updated to drop unneeded headers]
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use regmap-mmio instead of open-coding caching and register accessors.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
