A few driver and error handling fixes plus a fix to ensure that we
mute streams when we should. The Atmel trigger addition is a fix to
ensure that we do the correct sequence of interactions with the
hardware.
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Merge tag 'asoc-v3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A few driver and error handling fixes plus a fix to ensure that we
mute streams when we should. The Atmel trigger addition is a fix to
ensure that we do the correct sequence of interactions with the
hardware.
On the Dell machines with codec whose Subsystem Id is 0x10280610,
0x10280629 or 0x1028063e, no external microphone can be detected when
plugging a 3-ring headset. If we add "model=dell-headset-multi" for
the snd-hda-intel.ko, the problem will disappear.
The codecs on these machines belong to alc_269 family.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While enabling these machines, we found we would sometimes lose an
interrupt if we change hardware volume during playback, and that
disabling msi fixed this issue. (Losing the interrupt caused underruns
and crackling audio, as the one second timeout is usually bigger than
the period size.)
The machines were all machines from HP, running AMD Hudson controller,
and Realtek ALC282 codec.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1260225
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1986A codec is a pretty old codec and has really many hidden
restrictions. One of such is that each DAC is dedicated to certain
pin although there are possible connections. Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.
We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell machines with codec whose Subsystem Id is 0x10280624,
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/bugs/1259790
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF
outputs will have their IEC958 controls created starting from index 16
and the HDMI controls will be created starting from index 0.
However, HDMI simple_playback_build_controls() as used by old VIA and
NVIDIA codecs incorrectly requests the IEC958 controls to be created
with an S/PDIF type instead of HDMI.
In case the card has other codecs that have HDMI outputs, the controls
will be created with wrong index=16, causing them to e.g. be unreachable
by the ALSA "hdmi" alias.
Fix that by making simple_playback_build_controls() request controls
with HDMI indexes.
Not many cards have an affected configuration, but e.g. ASUS M3N78-VM
contains an integrated NVIDIA HDA "card" with:
- a VIA codec that has, among others, an S/PDIF pin incorrectly
labelled as an HDMI pin, and
- an NVIDIA MCP7x HDMI codec.
Reported-by: MysterX on #openelec
Tested-by: MysterX on #openelec
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> # 3.8+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat both negative and zero return values from clk_round_rate()
as errors. This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.
Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error; all other values will be
considered valid rates. The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat both negative and zero return values from clk_round_rate()
as errors. This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.
Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error. All other values will be
considered valid rates. The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have. This patch fixes
these holes. Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Inspiron 3045 machine (codec Subsystem Id: 0x10280628),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259437
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If snd_dmaengine_pcm_register()'s call to snd_soc_add_platform() fails,
all objects allocated during registration are leaked. Fix this by adding
error-handling code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If we update it here, the set_bias_level() of Codec driver won't be normally
called and we will then miss some essential procedures in set_bias_level() of
the Codec driver. Thus drop it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case,
"val" is never assigned to, but left uninitialized. The other case does
initialized it. Fix this by initializing val at the start of the
function, and only ever ORing into it.
Update the handling of "mask" so it works the same way for consistency.
Update tegra20_spdif.c to use the same code-style for consistency, even
though it doesn't happen to suffer from the same problem at present.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Reviewed-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes: 0f163546a7 ("ASoC: tegra: use regmap more directly")
Cc: <stable@vger.kernel.org>
Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD and VIA codecs had stereo mixer input enabled as default before
moving to the generic parser, and people think the lack of such a
regression. In this patch, the stereo mixer input is added back to
the input selection if no auto-mic is available, and if it's not
disabled explicitly via hint. This should satisfy most of demands,
i.e. stereo mix on desktop machines like what it worked before, and it
still keeps the new auto-mic feature on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes the hardware reports LPIB being advanced than POSBUF.
When this happens, the driver adjusts to a positive value by adding
the buffer size. Then the driver detects it as an error (greater than
the period size), and stops the LPIB delay account from this point
on.
When I took a close look at these conditions, the values shown are all
very small numbers, and it'd be better to just ignore these values
instead of discontinuing the LPIB delay correction.
In this patch, the driver checks a negative delay value and ignores if
it's a significantly small error. Currently the threshold is set to
64 frames, but could be smaller.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback mixing paths aren't initialized correctly at init
callback. Mostly this is harmless as codecs usually set the mute
state as default, but we still should make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have blindly assumed that all valid configurations should have
either analog or digital playback, but there can be capture-only
configurations. The parser shouldn't escape in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch skips the default depop delay before D3 for Haswell (10 ms) and
Valleyview2 (100 ms) display codec, to reduce codec suspend time.
The analog part of display audio is implemented in the external display. Some
displays have weak pop noise while others not when suspending, no matter there
is the default delay or not.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've tested the old Dell Vostro 131 with the latest generic parser
and it works just fine, and as a bonus we get better jack detection
features in userspace. Therefore this quirk can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Mikulas Patocka <mpatocka@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the
current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK,
which would cause the calculation result from DSPCLK_DIV invalid since bit
DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK
while the driver won't calculate it again for the current instance. In this
circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted
due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value.
So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for
calculation and then disables it afterward.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch add quirk for Acer Aspire E-572:
- fix external mic
- limit mic boost for internal mic with maximal noise level of -24dB
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MacBook Air 2,1 has a fairly different pin assignment from its brother
MBA 1,1, and yet another quirks are needed for pin 0x18 and 0x19,
similarly like what iMac 9,1 requires, in order to make the sound
working on it.
Reported-and-tested-by: Bruno Prémont <bonbons@linux-vserver.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change sam9x5 with wm8731 work in DSP A mode, this will fix the
left/right channel swap issue.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Tested-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to the SSC specifiation, it should be enabled after DMA is
enabled. So, add trigger operation to make sure the right sequence.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Tested-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
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Merge tag 'asoc-v3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
In the case of using jackpoll_ms instead of unsol events, the jack
was correctly detected, but ELD info was not refreshed on plug-in.
And without ELD info, no proper restriction of pcm, which can in turn
break sound output on some devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to remove the hp_automute_hook from alc283_fixup_chromebook.
It doesn't need this for other chrome os machine.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the CONFIG_PM explicity is undefined, let's convert to the
modern PM ops.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it
works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0.
So, fix LRP for DSP mode as the datesheet specification.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'.
And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce
codec suspend time.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create single model for HP.
The headset jack module was difference between other chrome book.
It need to manual control Mic jack detect.
Chrome OS loaded driver by models. Remove old assigned fixup table from
ALC269 fixup list entry.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By trial and error, I found this patch could work around an issue
where the headset mic would stop working if you switch between the
internal mic and the headset mic, and the internal mic was muted.
It still takes a second or two before the headset mic actually starts
working, but still better than nothing.
Information update from Kailang:
The verb was ADC digital mute(bit 6 default 1).
Switch internal mic and headset mic will run alc_headset_mode_default.
The coef index 0x11 will set to 0x0041.
Because headset mode was fixed type. It doesn't need to run
alc_determine_headset_type.
So, the value still keep 0x0041. ADC was muted.
BugLink: https://bugs.launchpad.net/bugs/1256840
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that AD1986A cannot manage the dynamic pin on/off for
auto-muting, but rather gets confused. Since each output has own amp,
let's use it instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ad_vmaster_eapd_hook() needs to handle the inverted EAPD case
properly, too. Otherwise the output gets broken on Lenovo N100 with
AD1986A codec.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z35HL laptop also needs the very same fix as the previous one
that was applied to ASUS W7J.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio devices tend to take long time for finishing the whole
probing procedure. In this patch, the time-consuming part of the
probing procedure, the codec probe and the rest initializations, are
moved in the work, so that they can be done asynchronously in parallel
with probes of other devices.
Since we already have this mechanism in the driver code for the
firmware and i915 request_symbol() stuff, we just need to enable it
always; the resultant patch even reduces more lines, which is an
additional bonus.
Credit goes to David Henningsson, who suggested this workaround.
Reported-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static checker found a possible array overflow in atmel/abdac.c:
static checker warning: "sound/atmel/abdac.c:373 set_sample_rates()
error: buffer overflow 'dac->rates' 6 <= 6"
This patch papers over the buggy point, by ensuring that dac->rates[]
update not overflowing the actual array size.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the probe of snd-hda-intel driver is deferred due to f/w loading
or the nested module loading, complete_all() should be also delayed
until the initialization really finished. Otherwise, vga-switcheroo
client would start switching before the actual init is done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
