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Dice quirk
In IEC 61883-6, one data block transfers one event. In ALSA, the event equals one PCM frame,
hence one data block transfers one PCM frame. But Dice has a quirk at higher sampling rate
(176.4/192.0 kHz) that one data block transfers two PCM frames.
Commit 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete
CIP_HI_DUALWIRE") moved some codes related to this quirk into Dice driver. But the commit
forgot to add arrangements for PCM period interrupts and DMA pointer updates. As a result, Dice
driver cannot work correctly at higher sampling rate.
This commit adds 'double_pcm_frames' parameter to amdtp structure for this quirk. When this
parameter is set, PCM period interrupts and DMA pointer updates occur at double speed than in
IEC 61883-6.
Reported-by: Daniel Robbins <drobbins@funtoo.org>
Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.16
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The channel mapping is initialized by amdtp_stream_set_parameters(), however
Dice driver set it before calling this function. Furthermore, the setting is
wrong because the index is the value of array, and vice versa.
This commit moves codes for channel mapping after the function and set it correctly.
Reported-by: Daniel Robbins <drobbins@funtoo.org>
Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # 3.16
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Acer Aspire V5 doesn't set up the pins correctly at the cold boot
while the pins are corrected after the warm reboot. This patch gives
the proper pin configs statically in the driver as a workaround.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=81561
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Right now we set 0 as the silence data for DSD_U8 and DSD_U16 formats,
but this is actually wrong. 0 is rather the most negative value.
Alternatively, we may take the repeating 0x69 pattern like ffmpeg
deploys.
Reference: https://ffmpeg.org/pipermail/ffmpeg-cvslog/2014-April/076427.html
Suggested-by: Alexander E. Patrakov <patrakov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_info_get_line() documents that its last parameter must be one
less than the buffer size, but this API design guarantees that
(literally) every caller gets it wrong.
Just change this parameter to have its obvious meaning.
Reported-by: Tommi Rantala <tt.rantala@gmail.com>
Cc: <stable@vger.kernel.org> # v2.2.26+
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Valleyview and Cherryview have the same behavior on display audio. So this patch
defines is_valleyview_plus() to include codecs for both Valleyview and its successor
Cherryview, and apply Valleyview fix-ups to Cherryview.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Both Haswell and Broadwell need set depop_delay to 0. So apply this
setting to haswell plus.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On some HP laptops, the mute led is controlled by codec gpio.
When some machine resume from s3/s4, the codec gpio data will be
cleared to 0 by BIOS:
Before suspend:
IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0
After resume:
IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0
To skip the AFG node to enter D3 can't fix this problem.
A workaround is to restore the gpio data when the system resume
back from s3/s4. It is safe even on the machines without this
problem.
BugLink: https://bugs.launchpad.net/bugs/1358116
Tested-by: Franz Hsieh <franz.hsieh@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ALC269 & co have many vendor-specific setups with COEF verbs.
However, some verbs seem specific to some codec versions and they
result in the codec stalling. Typically, such a case can be avoided
by checking the return value from reading a COEF. If the return value
is -1, it implies that the COEF is invalid, thus it shouldn't be
written.
This patch adds the invalid COEF checks in appropriate places
accessing ALC269 and its variants. The patch actually fixes the
resume problem on Acer AO725 laptop.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Tested-by: Francesco Muzio <muziofg@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.17
Nothing too exciting here, a bunch of driver fixes that came along since
the initial pull request but none that really stand our and a warning
fix in the core.
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'asoc/fix/fsl-esai', 'asoc/fix/intel', 'asoc/fix/mcasp' and 'asoc/fix/pxa' into asoc-linus
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Conexnat HD-audio driver has a workaround for cx5051 (aka CX20561)
chip to add fake mute controls to each amp (commit 3868137e). This
implies the minimum-as-mute TLV bit in TLV for each corresponding
control. Meanwhile we build the virtual master from these, but the
TLV bit is missing, even though the slaves have it.
This patch simply adds the missing TLV_DB_SCALE_MUTE bit for vmaster,
as already done in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This mode is unsupported, as the DMA controller can't do zero-padding
of samples.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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This reverts commit a603c8ee526f5ea9ad9b40710308766299ad8a69.
fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask().
fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit
to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled
bit to 0.
For esai when the bit value is 1, the slot is enabled, when the bit value is 0,
the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will
work abnormally. So revert this patch, make the esai use default function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The implicit BLCK divider setting was broken by "ASoC: mcasp: don't
override bclk divider if it was provided by the machine"-patch. After
the BCLK divider is implicitly set for the first time the
mcasp->bclk_div gets a non zero value and the implicit setting is
"turned off".
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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TDM slot length was set same as word length, regardless of the value
received in set_tdm_slot. This patch sets the TDM slot length correctly
as received in set_tdm_slot DAI callback
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The source type should come before the direction specifier according to
ControlNames.txt.
Signed-off-by: Mark Brown <broonie@linaro.org>
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If soc_dapm_read() fails, reg_val will be uninitialized, and bogus
values will be written later:
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double':
sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized]
unsigned int reg_val, val;
^
Return early on error to fix this.
Introduced by commit ce0fc93ae56e2ba50ff8c220d69e4e860e889320 ("ASoC:
Add DAPM support at the component level").
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
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There is no need to restore and restart PCM streams in case ADSP didn't
reach reset and power off state during system suspend/resume cycle. In that
case stream is still active but paused and firmware doesn't allow allocating
a new stream before paused stream is freed.
ADSP remains active in case suspend sequence didn't go to suspend_late
stage. This can happen when either suspend sequence is aborted by a wakeup
or by letting only devices suspend by "echo devices >/sys/power/pm_test".
Currently stream restoring fails in these suspend cases. Fix this by adding
a flag that indicates is complete stream reinitialization needed or is it
enough to resume paused stream. Flag is set when we know that ADSP reached
suspend_late.
Initial fix to this issue came from Fang Yang. I modified it a little and
forward ported it to top of two other suspend/resume patches from me.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Cc: yang fang <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Remove sst_byt_pcm_dev_resume() and move waiting of firmware boot into
sst_byt_pcm_dev_resume_early(). Now suspend_late and resume_early phases are
in sync with each other so that we know that ADSP was put into reset and was
unpowered after suspend_late and is ready to resume IO after resume_early
during resume stage in sst_byt_pcm_trigger().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Merge DSP reset and cleanup sequence in sst_byt_pcm_dev_suspend_noirq()
into sst_byt_pcm_dev_suspend_late(). First their order was wrong by first
unloading firmware modules in suspend_late and then taking DSP into reset
in suspend_noirq. Second ACPI has put device into OFF state already during
suspend_late so trying to reset the DSP is a no-op at suspend_noirq stage.
Fix these by moving DSP reset and cleanup into
sst_byt_pcm_dev_suspend_late() before firmware unloading.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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CA0132 driver tries to reload the firmware at resume. Usually this
works since the firmware loader core caches the firmware contents by
itself. However, if the driver failed to load the firmwares
(e.g. missing files), reloading the firmware at resume goes through
the actual file loading code path, and triggers a kernel WARNING like:
WARNING: CPU: 10 PID:11371 at drivers/base/firmware_class.c:1105 _request_firmware+0x9ab/0x9d0()
For avoiding this situation, this patch makes CA0132 skipping the f/w
loading at resume when it failed at probe time.
Reported-and-tested-by: Janek Kozicki <cosurgi@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If nid 0x15 (Headphone Playback Switch) is in D3 and headphones are
plugged in when the laptop reboots, a pop noise is generated.
Prevent this by keeping nid 0x15 in D0 when headphones are plugged in.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If the laptop is powered on with a jack plugged in, independently on what
is plugged, the jack is treated as a microphone jack.
Initialize the capture source so that by default jacks are treated as
headphones jacks. This will also prevent pop noises on boot in case
headphones are plugged in since setting/unsetting mic-in as input source
causes a pop noise.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The BOSS ME-25 turns out not to have any useful descriptors in its MIDI
interface, so its needs a quirk entry after all.
Reported-and-tested-by: Kees van Veen <kees.vanveen@gmail.com>
Fixes: 8e5ced83dd1c ("ALSA: usb-audio: remove superfluous Roland quirks")
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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CMI8888 codec chip has a boost amp (only) on the headphone pin, and
this confuses the generic parser, which tends to pick up the most
outside amp. This results in the wrong volume setup, as the driver
complains like:
hda_codec: Mismatching dB step for vmaster slave (-100!=1000)
For avoiding this problem, rule out the amp on NID 0x10 and create
"Headphone Amp" volume control manually instead.
Note that this patch still doesn't fix all problems yet. The sound
output from the line out seems still too low. It will be fixed in
another patch (hopefully).
Reported-and-tested-by: Vincent Lejeune <vljn@ovi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS Phoebus with CMI8888 HD-audio chip (PCI id 13f6:5011) doesn't
work with HD-audio driver as is because of some weird nature. For
making DMA properly working, we need to disable MSI. The position
report buffer doesn't work, thus we need to force reading LPIB
instead. And yet, the codec CORB/RIRB communication gives errors
unless we disable the snooping (caching).
In this patch, all these workarounds are added as a quirk for the
device. The HD-audio *codec* chip needs yet another workaround, but
it'll be provided in the succeeding patch.
Reported-and-tested-by: Vincent Lejeune <vljn@ovi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It will be recording voice delay for resume back recording for Headset Mic.
This alc286 will quickly open Headset Mic, to prevent avoid recording files are missing.
The issue was fixed. This is follow ALC286 programing guide.
[fix build error, add static and renamed the function by tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Update the initial Baytrail ADSP firmware file name with the one that is now
in linux-firmware.git. Please see linux-firmware.git commit 7551a3a78453
("fw_sst_0f28: Add firmware for Intel Baytrail SST DSP").
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedefs for wanc_info and
wavnc_port_info.
A simplified version of the Coccinelle semantic patch that finds the case is:
@tn@
identifier i;
type td;
@@
-typedef
struct i { ... }
-td
;
@@
type tn.td;
identifier tn.i;
@@
-td
+ struct i
Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedef for uart401_devc.
The following Coccinelle semantic patch detects the case.
@tn@
identifier i;
type td;
@@
-typedef
struct i { ... }
-td
;
@@
type tn.td;
identifier tn.i;
@@
-td
+ struct i
Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/usb/card.c registers USB suspend and resume but did not previously
kill the input URBs. This means that USB MIDI devices left open across
suspend/resume had non-functional input (output still usually worked,
but it looks like that is another issue). Before this change, we would
get ESHUTDOWN for each of the input URBs at suspend time, killing input.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedefs for pss_mixerdata
and pss_confdata.
The following Coccinelle semantic patch is used to make the change.
@tn@
identifier i;
type td;
@@
-typedef
struct i { ... }
-td
;
@@
type tn.td;
identifier tn.i;
@@
-td
+ struct i
Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This typedef is unnecessary and should just be removed as they are
never used.
The following Coccinelle semantic patch detects the case.
@tn@
identifier i;
type td;
@@
-typedef
struct i { ... }
-td
;
@@
type tn.td;
identifier tn.i;
@@
-td
+ struct i
Signed-off-by: Himangi Saraogi <himangi774@gmail.com>
Acked-by: Julia Lawall <julia.lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use %d for loop counter and %X for device capabilities. This is a
supplemental patch for Hans Wennborg's patch.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.17
This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC. A big thanks to Benoit and Misael for their work on
this.
Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz.
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah.
- More componentisation work from Lars-Peter Clausen.
- The remainder of the conversions of CODEC drivers to params_width()
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
TAS2552.
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers.
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'asoc/topic/wm0010', 'asoc/topic/wm8904' and 'asoc/topic/wm8962' into asoc-next
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'asoc/topic/tlv320aic31xx' and 'asoc/topic/tlv320aic32x4' into asoc-next
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'asoc/topic/spdif', 'asoc/topic/tas2552' and 'asoc/topic/tas5086' into asoc-next
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'asoc/topic/s6000', 'asoc/topic/samsung' and 'asoc/topic/sh-fsi' into asoc-next
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'asoc/topic/rt286', 'asoc/topic/rt5640' and 'asoc/topic/rt5645' into asoc-next
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'asoc/topic/pxa' into asoc-next
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'asoc/topic/max98090' and 'asoc/topic/mc13783' into asoc-next
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'asoc/topic/fsl-spdif' and 'asoc/topic/imx-audmux' into asoc-next
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'asoc/topic/cs42xx8', 'asoc/topic/cx20442' and 'asoc/topic/davinci' into asoc-next
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