Commit Graph

14294 Commits

Author SHA1 Message Date
Padmavathi Venna
40476f6189 ASoC: samsung: Add DT support for i2s
Add support for device based discovery.

Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-29 12:47:46 +08:00
Padmavathi Venna
2d77828d99 ASoC: Samsung: Add I2S S/W RST in startup function
I2S module need to be reset after S2R. Keeping the S/W rst
control part in resume didn't help in playing audio after resume.
So this patch adds S/W RST control part in startup function which
gets triggered for every new audio stream playback.

Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: R. Chandrasekar <rcsekar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-26 15:24:51 +08:00
Padmavathi Venna
7c62eebbf7 ASoC: samsung: Rename samsung i2s secondary device name
All Samsung SoCs has max 3 i2s controllers. So the i2s secondary fifo
interface device id was named as samsung-i2s.4. Renaming this to
"samsung-i2s-sec" to support device tree in i2s driver.

Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-24 14:40:13 +08:00
Takashi Iwai
6f54c36132 ALSA: hda/hdmi - Work around "alsactl restore" errors
When "alsactl restore" is performed on HDMI codecs, it tries to
restore the channel map value since the channel map controls are
writable.  But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream
is assigned yet, and this results in an error message from alsactl.
Although the error is harmless, it's certainly ugly and can be
regarded as a regression.

As a workaround, this patch changes the return code in such a case to
be zero for making others happy.  (A slight excuse is: when the chmap
is changed through the proper alsa-lib API, the PCM status is checked
there anyway, so we don't have to be too strict in the kernel side.)

Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 14:55:16 +01:00
Eldad Zack
39e95156b9 ALSA: usb-audio: selector map for M-Audio FT C400
Add names of the clock sources for the M-Audio Fast Track
C400.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:11 +01:00
Eldad Zack
83e3acd494 ALSA: usb-audio: M-Audio FT C400 skip packet quirk
Attain constant real-world latency by skipping 16 data packets.
The number of packets to be skipped was found by trial and error.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:03 +01:00
Eldad Zack
2aad272b3f ALSA: usb-audio: correct M-Audio C400 clock source quirk
Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.

Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:05:57 +01:00
David Henningsson
b98ae2729d ALSA: usb - fix race in creation of M-Audio Fast track pro driver
A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.

However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?

BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:03:03 +01:00
Takashi Iwai
31be5425d7 ALSA: usb-audio: Fix NULL dereference by access to non-existing substream
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP.  But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.

This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.

Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-11 11:12:17 +01:00
Takashi Iwai
c18ab0bac4 ASoC: Fixes for v3.8
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
 just hadn't worked before, nobody noticed due to lack of use.
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Merge tag 'asoc-fix-3.8-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.8

Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
2013-01-10 17:41:54 +01:00
Mark Brown
49a170bcf2 Merge remote-tracking branch 'asoc/fix/wm5100' into tmp 2013-01-10 12:22:30 +00:00
Mark Brown
921c038d87 Merge remote-tracking branch 'asoc/fix/wm2200' into tmp 2013-01-10 12:22:29 +00:00
Mark Brown
28f2675db8 Merge remote-tracking branch 'asoc/fix/wm2000' into tmp 2013-01-10 12:22:26 +00:00
Mark Brown
92a9d1524e Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp 2013-01-10 12:22:25 +00:00
Mark Brown
a883eae513 Merge remote-tracking branch 'asoc/fix/sta529' into tmp 2013-01-10 12:22:22 +00:00
Mark Brown
fd2eab87a2 Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp 2013-01-10 12:22:17 +00:00
Mark Brown
87fee06c5b Merge remote-tracking branch 'asoc/fix/pxa' into tmp 2013-01-10 12:22:16 +00:00
Mark Brown
c31b71de6f Merge remote-tracking branch 'asoc/fix/lm49453' into tmp 2013-01-10 12:22:15 +00:00
Mark Brown
fa17cb4a02 Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp 2013-01-10 12:22:14 +00:00
Mark Brown
587691ea39 Merge remote-tracking branch 'asoc/fix/cs4271' into tmp 2013-01-10 12:22:11 +00:00
Mark Brown
a18a31a161 Merge remote-tracking branch 'asoc/fix/core' into tmp 2013-01-10 12:21:50 +00:00
Mark Brown
ae1abb0c3b Merge remote-tracking branch 'asoc/fix/arizona' into tmp 2013-01-10 12:21:42 +00:00
Kailang Yang
065380f088 ALSA: hda - Add support of new codec ALC284
Added the support for a new codec ALC284, which is compatible with
ALC269.  Also add more codec variants to handle the SSID check
properly.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:27:46 +01:00
Sachin Kamat
e8e7da23c9 ALSA: usb-audio: Make ebox44_table static
Fixes the following sparse warning:
sound/usb/mixer_quirks.c:1209:23: warning:
symbol 'ebox44_table' was not declared. Should it be static?

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:22:25 +01:00
Andre Schramm
56bde0f328 ALSA: hdspm - Fix wordclock status on AES32
Use correct bitmask for AES32 cards to determine wordclock lock state,
add missing bitmask for sync check and make output of the corresponding
control and /proc coherent.

Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 16:59:24 +01:00
David Henningsson
7ed4165e2d Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
This reverts commit 697c373e34.

The original patch was meant to remove clicking, but in fact caused even
more clicking instead.

Thanks to c4pp4 for doing most of the work with this bug.

BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:03:38 +01:00
Takashi Iwai
d7dab4dbbb ALSA: hda - Disable runtime D3 for Intel CPT & co
We've got a few bug reports that the runtime D3 results in the dead
HD-audio controller.  It seems that the problem is in a deeper level
than the sound driver itself, so as a temporal solution, disable the
feature for these controllers again.

Reported-and-tested-by: Vincent Blut <vincent.debian@free.fr>
Reported-and-tested-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org> [v3.7]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:00:08 +01:00
Mike Dunn
3b4bc7bccc ALSA: pxa27x: fix ac97 warm reset
This patch fixes some code that implements a work-around to a hardware bug in
the ac97 controller on the pxa27x.  A bug in the controller's warm reset
functionality requires that the mfp used by the controller as the AC97_nRESET
line be temporarily reconfigured as a generic output gpio (AF0) and manually
held high for the duration of the warm reset cycle.  This is what was done in
the original code, but it was broken long ago by commit fb1bf8cd
    ([ARM] pxa: introduce processor specific pxa27x_assert_ac97reset())
which changed the mfp to a GPIO input instead of a high output.

The fix requires the ac97 controller to obtain the gpio via gpio_request_one(),
with arguments that configure the gpio as an output initially driven high.

Tested on a palm treo 680 machine.  Reportedly, this broken code only prevents a
warm reset on hardware that lacks a pull-up on the line, which appears to be the
case for me.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-08 11:27:35 +00:00
Mike Dunn
41b645c862 ALSA: pxa27x: fix ac97 cold reset
Cold reset on the pxa27x currently fails and

     pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44)

appears in the kernel log.  Through trial-and-error (the pxa270 developer's
manual is mostly incoherent on the topic of ac97 reset), I got cold reset to
complete by setting the WARM_RST bit in the GCR register (and later noticed that
pxa3xx does this for cold reset as well).  Also, a timeout loop is needed to
wait for the reset to complete.

Tested on a palm treo 680 machine.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Acked-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-08 11:27:27 +00:00
Mark Brown
a76fefab5c ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
Otherwise we won't run correctly on systems that require this for larger
data transfers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-07 19:13:35 +00:00
David Henningsson
6d3cd5d444 ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
The mute LED is in this case connected to the Mic1 VREF.

The machine also exposes the following string in BIOS:
"HP_Mute_LED_0_A", so if more machines are coming, it probably
makes sense to try to do something more generic, like for the
IDT codec.

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1096789
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-07 17:29:55 +01:00
Nickolai Zeldovich
61ed1dca16 ALSA: au88x0: fix incorrect left shift
vortex_wt_setdsout performs bit-negation on the bit position (wt&0x1f)
rather than on the resulting bitmask.  This code is never actually
invoked (vortex_wt_setdsout is always called with en=1), so this does
not currently cause any problem, and this patch is simply cleanup.

Signed-off-by: Nickolai Zeldovich <nickolai@csail.mit.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-07 09:33:56 +01:00
Mark Brown
267f8fa2e1 ASoC: wm2000: Fix sense of speech clarity enable
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 21:19:42 +00:00
Mark Brown
5f960294e2 ASoC: wm5100: Remove DSP B and left justified formats
These are not supported

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 21:06:08 +00:00
Mark Brown
d71753e22b ASoC: arizona: Remove DSP B and left justified AIF modes
These are not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 11:33:22 +00:00
Mark Brown
0cc411b934 ASoC: wm2200: Remove DSP B and left justified AIF modes
These are not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 11:31:57 +00:00
Asim Kadav
dc30a43690 sound: oss/pas2: Fix possible access out of array
Added a fix for hardware dependence bug where a sound card failure
should not result in reading beyond array memory index.

Signed-off-by: Asim Kadav <kadav@cs.wisc.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 10:38:27 +01:00
Damien Zammit
b7b435e81b ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirk
This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
   "bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
   invalid midi endpoint.

It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 09:53:17 +01:00
Alexander Schremmer
8f7f3ab15e ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirks
Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:

 no or invalid class specific endpoint descriptor

Now it still prints these messages on hotplug:

 snd-usb-audio: probe of ...:1.0 failed with error -5
 snd-usb-audio: probe of ...:1.2 failed with error -5
 snd-usb-audio: probe of ...:1.3 failed with error -5

But the device works correctly, including the HID support.

The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.

Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:26:48 +01:00
David Henningsson
c86c2d440c ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrol
The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted,
but "Mute-LED Mode" being "On" refers to the LED being on, not the
volume being on.
Therefore "On" and "Off" should be switched.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:22:34 +01:00
Mark Brown
1b8d52e63c ASoC: wm5102: Improve speaker enable performance
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:08:42 +00:00
Chuansheng Liu
d3bf156125 ASoC: core: fix the memory leak in case of remove_aux_dev()
When probing aux_dev, initializing is as below:
device_initialize()
device_add()

So when remove aux_dev, we need do as below:
device_del()
device_put()
Otherwise, the rtd_release() will not be called.

So here using device_unregister() to replace device_del(),
like the action in soc_remove_link_dais().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:14:43 +00:00
Chuansheng Liu
865df9cb12 ASoC: core: fix the memory leak in case of device_add() failure
After called device_initialize(), even device_add() returns
error, we still need use the put_device() to release the reference
to call rtd_release(), which will do the free() action.

Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:14:43 +00:00
Axel Lin
3271a4fc7d ASoC: cs42l52: Catch no-match case in cs42l52_get_clk
In the case of no-match, return -EINVAL instead of 0.

Since we assign i to ret in the for loop, ret always less than
ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 16:02:19 +00:00
MR.Swami.Reddy@ti.com
9dc754dfa7 ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
Update lm49453_reg_defs values as per LM49453 HW revision-B

Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:45:10 +00:00
MR.Swami.Reddy@ti.com
88ac43924b ASoC: lm49453: Fix adc, mic and sidetone volume ranges
Add adc, mic, sidetone volume ranges and appropriately added the controls.
Fix the DAC HP/EP/LS/LO/HA maximum gain values.

Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com>
Tested-by: Vinod Koul <vinod.koul@intel.com>

--
 sound/soc/codecs/lm49453.c |   43 ++++++++++++++++++++++++-------------------
 1 files changed, 24 insertions(+), 19 deletions(-)
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:43:56 +00:00
Mark Brown
a8c02db029 ASoC: arizona: Correct FLL source definitions
The FLL source constants were numbered as a simple enumeration but were
being used in the code as direct values to be written to the registers.
Renumber the constants to reflect the usage.

Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-24 15:41:44 +00:00
Axel Lin
7110a287ff ASoC: arizona: Do proper shift for setting AIF rate
ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */
Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-24 15:41:44 +00:00
Axel Lin
ec20f2f8d3 ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
The mode variable is either 0 or 1.
To update mode setting, the mask should be BIT(0) rather than BIT(1).

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Omair M. Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:32:28 +00:00
Fabio Estevam
b50684da6c ASoC: sgtl5000: Fix maximum value for microphone gain
sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3.

From Eric Nelson:
"We also found that for the microphones we have here (commodity PC boom mics) a
default value of 2 for the gain gives the best results."

So change the default microphone gain as well.

Signed-off-by: Eric Nelson <eric.nelson@boundarydevices.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:31:25 +00:00