If an endpoint in use, its associated URBs should not be
deactivated.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The only call site for deactivate_endpoints() at snd_usb_hw_free().
The return value is not checked there, as it is irrelevant if it
fails on hw_free.
This patch moves the deactivation of the endpoints directly into
snd_usb_hw_free().
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Since the format is not actually used in sync_ep_set_params(),
there is no need to pass it down.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds LED support for the Native Instruments Maschine
Controller. It adds ALSA controls for dimming the LEDs of all
buttons and the backlight of the two displays.
Signed-off-by: NHannes Gräuler <hgraeule@uos.de>
Acked-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Calculating frame bytes can be replaced with inline function in
include/sound/pcm.h.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch changes the way URBs are allocated and their sizes are
determined for PCM playback in the snd-usb-audio driver.  Currently
the driver allocates too few URBs for endpoints that don't use
implicit sync, making underruns more likely to occur.  This may be a
holdover from before I/O delays could be measured accurately; in any
case, it is no longer necessary.

The patch allocates as many URBs as possible, subject to four
limitations:

	The total number of URBs for the endpoint is not allowed to
	exceed MAX_URBS (which the patch increases from 8 to 12).

	The total number of packets per URB is not allowed to exceed
	MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is
	decreased from 20 to 6.

	The total duration of queued data is not allowed to exceed
	MAX_QUEUE, which is decreased from 24 ms to 18 ms.

	The total number of ALSA frames in the output queue is not
	allowed to exceed the ALSA buffer size.

The last requirement is the hardest to implement.  Currently the
number of URBs needed to fill a buffer cannot be determined in
advance, because a buffer contains a fixed number of frames whereas
the number of frames in an URB varies to match shifts in the device's
clock rate.  To solve this problem, the patch changes the logic for
deciding how many packets an URB should contain.  Rather than using as
many as possible without exceeding an ALSA period boundary, now the
driver uses only as many packets as needed to transfer a predetermined
number of frames.  As a result, unless the device's clock has an
exceedingly variable rate, the number of URBs making up each period
(and hence each buffer) will remain constant.

The overall effect of the patch is that playback works better in
low-latency settings.  The user can still specify values for
frames/period and periods/buffer that exceed the capabilities of the
hardware, of course.  But for values that are within those
capabilities, the performance will be improved.  For example, testing
shows that a high-speed device can handle 32 frames/period and 3
periods/buffer at 48 KHz, whereas the current driver starts to get
glitchy at 64 frames/period and 2 periods/buffer.

A side effect of these changes is that the "nrpacks" module parameter
is no longer used.  The patch removes it.
Signed-off-by: NAlan Stern <stern@rowland.harvard.edu>
CC: Clemens Ladisch <clemens@ladisch.de>
Tested-by: NDaniel Mack <zonque@gmail.com>
Tested-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BIOS can mark a pin as "no physical connection" if the port is used by an
integrated display which is not audio capable. And audio driver will overlook
such pins.

On Haswell, such a disconneted pin will keep muted and connected to the 1st
converter by default. But if the 1st convertor is assigned to a connected pin
for audio streaming. The muted disconnected pin can make the connected pin
no sound output.

So this patch avoids using assigned converters for all unused pins for Haswell,
including the disconected pins.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Reviewed-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Convert 0 to false and 1 to true when assigning values to bool
variables. Inspired by commit 3db1cd5c.

The simplified semantic patch that find this problem is as
follows (http://coccinelle.lip6.fr/):

@@
bool b;
@@
(
-b = 0
+b = false
|
-b = 1
+b = true
)
Signed-off-by: NPeter Senna Tschudin <peter.senna@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Currently we assume that userspace will shut down the compressed stream
correctly. However, if userspcae dies (e.g. cplay & ctrl-C) we dont
stop the stream before freeing it.

This now checks that the stream is stopped before freeing.
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

snd_unregister_device() should return the device type and not stream
direction.
Signed-off-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Tested-by: NVinod Koul <vinod.koul@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

After the last architecture switched to generic hard irqs the config
options HAVE_GENERIC_HARDIRQS & GENERIC_HARDIRQS and the related code
for !CONFIG_GENERIC_HARDIRQS can be removed.
Signed-off-by: NMartin Schwidefsky <schwidefsky@de.ibm.com>

'break' after return statement is not necessary.
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

'break' after a return statement is redundant. Remove it.
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

'codec_send_command' is used only in this file. Make it static.
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Local symbols used only in this file are made static.
Signed-off-by: NSachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

These ->put() functions are called from snd_ctl_elem_write() with user
supplied data.  snd_asihpi_tuner_band_put() is missing a limit check and
the check in snd_asihpi_clksrc_put() can underflow.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The MC13783 Chip Errata, Rev. 4 says, that depending on SPI clock
and main audio clock speed, the Audio Codec or Stereo DAC do sometimes
not start when programmed to do so. This is due to an internal clock
timing issue related to the loading of the SPI bits into the audio block.

On an i.MX27 based system, this issue lead to switched audio channels under
certain circumstances: RTC + Touch + Audio are used and loaded at startup.

The mentioned workaround of writing registers 40 and 41 two times is implemented
here.
Signed-off-by: NSteffen Trumtrar <s.trumtrar@pengutronix.de>
Cc: stable@vger.kernel.org
Signed-off-by: NMark Brown <broonie@linaro.org>

it should be *USE*, not *USB*
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

MacBook 6,1 and 6,2 have a CS4208 codec instead of CS4206/CS4207 on
the former models.  Most of functions work fine as is, except for the
silent speaker output.  After debugging sessions, it turned out that
the machine needs to set GPIO 0 for the speaker amp.

This patch adds the basic support for CS4208 and the fixup for these
MacBooks.  Basically the codec works just with the generic parser.
For re-using the existing GPIO amp code and init/free callbacks, a few
places have been changed so that CS4206/4207-specific codes (errata,
etc) won't hit with CS4208.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60811Reported-and-tested-by: NImre Kaloz <kaloz@openwrt.org>
Reported-and-tested-by: NIan Munsie <darkstarsword@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Toshiba Satellite C870 shows interrupt problems occasionally when
certain mixer controls like "Mic Switch" is toggled.  This seems
worked around by not using MSI.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=833585
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The S/PDIF driver needs regmap so select it to make sure it gets
included in the build.
Reported-by: NFengguang Wu <fengguang.wu@intel.com>
Acked-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

When Gfx driver reconnects a port and transcoder, the pin amplifier will
be muted. To enable sound, the pin amp need to be unmuted.

This patch
- moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe().
  So if port:transcoder reconnection happens during stream playback, the ELDV
  unsol event can stil trigger pin's amp unmuting when re-setting up audio
  info frame.

- remove reading pin amp status before unmuting for speed-up, since pin amp
  should always be unmuted.

- rename haswell_verify_pin_D0() to haswell_verify_D0(), since the convertor
  power state is also fixed here.

This patch is mostly based on suggestion of David Henningsson.

Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

To apply Haswell specific fixings, this patch defines is_haswell() to check
whether a display audio codec is Haswell, to avoid explicitly checking Haswell
vendor ID everywhere.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ASUS TX300 has a built-in speaker in the tablet part and in the dock
part, and the tablet speaker is supposed to be unused while the
machine is docked.  The current HD-audio driver, however, doesn't
support the dock speaker, partly because BIOS doesn't set up the pin
for the corresponding output.

But, not only the missing pin config, also the missing unsol event
handling is another issue.  Otherwise the automatic switching via
dock/undock won't work.

Through debugging sessions, we found out that the dock speaker pin is
NID 0x1b, and it generates an unsol event at docking/undocking, the
docking state can be inquired via the normal pin detection verb.
Also, it's turned out that GPIO 2 is needed as an amp.  So, all
materials are ready to cook.

This patch provides the basic dock speaker support with TX300:
- The dock speaker is turned on/off via "Dock Speaker" mixer mute.
- The dock speaker is automatically muted when docked.  This is
  independently from the mixer mute switch, just like the headphone
  auto-mute function.

The implementation is a bit tricky.  Since we want to handle it as a
secondary speaker, we set it up a pin as a speaker with a jack
detection.  Then, the fixup function registers the own unsol callback
for this pin because the standard automute can't handle the thing like
a "speaker jack".  In the own automute hook, we apply the mute of the
tablet speaker in addition by checking the dock state.

Also, the speaker control names are slightly shuffled because the
generic parser doesn't give good names but blindly assumes a bass
speaker as a secondary speaker.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59791Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The compatible string of the kirkwood-i2s driver was chosen as
"marvell,mvebu-audio". Using such a compatible string is not a good
idea, since "mvebu" is the name of a large family of SOCs, in which
new, unknown SOCs will be coming in the future. It is therefore
impossible to know what will be evolutions of this hardware block in
the next generations of the SOCs. For this reason, the recommandation
for compatible strings of on-SOCs devices has always been to use the
name of the oldest SOC that has the hardware block. New SOCs that have
an exactly compatible hardware block can reference it using the same
compatible string. See [1], [2] and [3] for various cases were this
suggestion was made, including from Rob Herring, a Device Tree binding
maintainer.

As an example, there are already small differences between current
generations:

 * On Kirkwood, only one interrupt is used for audio.
 * On Dove, two interrupts are used, one for audio data and one for
   error reporting.

In the near future, I'll be adding audio support to Armada 370, which
allows has the same hardware block (but maybe with minor variants).

Therefore, this patch changes the driver to accept
"marvell,kirkwood-audio" and "marvell,dove-audio" as compatible
strings instead of the too-generic "marvell,mvebu-audio". The reason
for the two different compatible strings is the difference in the
number of interrupts used by the two SOCs for audio.

This Device Tree binding has never been part of a Linux kernel stable
release so far, so it can be changed now without breaking backward
compatibility.

[1] http://lists.infradead.org/pipermail/linux-mtd/2012-March/040417.html
[2] http://lists.infradead.org/pipermail/linux-arm-kernel/2013-April/161065.html
[3] http://lists.infradead.org/pipermail/linux-arm-kernel/2012-March/087702.htmlSigned-off-by: NThomas Petazzoni <thomas.petazzoni@free-electrons.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

As once the error interrupt is triggered, it can not be cleared.
So, disable it.

No side effect found while testing on sama5d3xek and at91sam9x5ek
boards.
Signed-off-by: NBo Shen <voice.shen@atmel.com>
Acked-by: NNicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Booting a mx51babbage board with a non-dt kernel leads to the following crash:

Unable to handle kernel NULL pointer dereference at virtual address 0000001c
pgd = 80004000
[0000001c] *pgd=00000000
Internal error: Oops: 5 [#1] SMP ARM
Modules linked in:
CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287
task: 9f860000 ti: 9f862000 task.ti: 9f862000
PC is at of_get_next_available_child+0x5c/0x68
LR is at of_get_next_available_child+0x1c/0x68
pc : [<8043ea58>]    lr : [<8043ea18>]    psr: 60000193
sp : 9f863d58  ip : 00000000  fp : 9f863d74
r10: 9f89a010  r9 : 9f862000  r8 : 807bb26c
r7 : 80615d5c  r6 : 00000000  r5 : 60000113  r4 : 00000000
r3 : 00000000  r2 : 808770a4  r1 : 00000011  r0 : 60000113
Flags: nZCv  IRQs off  FIQs on  Mode SVC_32  ISA ARM  Segment kernel
Control: 10c5387d  Table: 90004019  DAC: 00000017
Process swapper/0 (pid: 1, stack limit = 0x9f862240)
Stack: (0x9f863d58 to 0x9f864000)

This is caused by commit 8548a464 (ASoC: imx-audmux: Read default
configuration from devicetree).

In order to fix this, add a check for 'of_id' so that
imx_audmux_parse_dt_defaults() only gets called when a dt kernel is running.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

When the transcoder:port mapping on Haswell HDMI/DP audio is changed
during the stream playback, the sound gets lost.  Typically this
problem is seen when the user switches the graphics mode from eDP+DP
to DP-only configuration, where CRTC 1 is used for DP in the former
while CRTC 0 is used for the latter.

The graphics controller notifies the change via the normal ELD update
procedure, so we get the intrinsic event.  For enabling the sound
again, the HDMI audio driver needs to reset the pin and set up the
audio infoframe again.

This patch achieves it by:
- keep the current status of channels and info frame setup in per_pin
  struct,
- check the reconnection in the intrinsic event handler,
- reset the pin and the re-invoke hdmi_setup_audio_infoframe()
  accordingly.

The hdmi_setup_audio_infoframe() function has been changed, too, so
that it can be invoked without passing the substream instance.

The patch is mostly based on the work by Mengdong Lin.

Cc: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

hdmi_channel_allocation() tries to find a HDMI channel allocation that
matches the number channels in the playback stream and contains only
speakers that the HDMI sink has reported as available via EDID. If no
such allocation is found, 0 (stereo audio) is used.

Using CA 0 causes the audio causes the sink to discard everything except
the first two channels (front left and front right).

However, the sink may be capable of receiving more channels than it has
speakers (and then perform downmix or discard the extra channels), in
which case it is preferable to use a CA that contains extra channels
than to use CA 0 which discards all the non-stereo channels.

Additionally, it seems that HBR (HD) passthrough output does not work on
Intel HDMI codecs when CA is set to 0 (possibly the codec zeroes
channels not present in CA). This happens with all receivers that report
a 5.1 speaker mask since a HBR stream is carried on 8 channels to the
codec.

Add a fallback in the CA selection so that the CA channel count at least
matches the stream channel count, even if the stream contains channels
not present in the sink speaker descriptor.

Thanks to GrimGriefer at OpenELEC forums for discovering that changing
the sink speaker mask allowed HBR output.

Reported-by: GrimGriefer
Reported-by: Ashecrow
Reported-by: NFrank Zafka <kafkaesque1978@gmail.com>
Reported-by: NPeter Frühberger <fritsch@xbmc.org>
Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On a mx28 board, running "aplay -l" and "arecord -l" results in the following:

$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0

$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0

,which is not correct because we got a capture device listed in aplay and a
playback device listed in arecord.

On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each
one of them are unidirectional.

Allow to specify a dai link as 'playback_only' or 'capture_only', which suits
well for this case.

After this change we can correctly report the capabilities as follows:

$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0

$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Also tested playback and capture on the mx28evk board.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.

The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Signed-off-by: NMark Brown <broonie@linaro.org>
Acked-by: NNicolin Chen <B42378@freescale.com>

The ssm2602 uses regmap for caching not soc-cache, so we need to use
regcache_sync() instead of snd_soc_cache_sync().
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

There is a blank space missing between ':=' and 'imx-spdif.o', thus add it.
Signed-off-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Add wrapping '\n' for dev_dbg() in fsl_spdif.c
Signed-off-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Signed-off-by: NBard Liao <bardliao@realtek.com>
Tested-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

We need to make sure that the control's cached value is initialized to the same
value as the control's widget->on_val. Otherwise updates might be lost.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the
i.mx case.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Acked-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The legacy S3C-DMA API required every period of a cyclic buffer to be
queued separately. After conversion of Samsung ASoC to Samsung DMA
wrappers somebody made an assumption that the same is needed for DMA
engine API, which is not true.

In effect, Samsung ASoC DMA code was queuing the whole cyclic buffer
multiple times with a shift of one period per iteration, leading to:
  a) severe memory waste - up to 13x times more DMA transfer descriptors
     are allocated than needed,
  b) possible memory corruption, because further cyclic buffers were out
     of the original buffers, due to the offset.

This patch fixes this problem by making the legacy S3C-DMA API use the
same semantics as DMA engine (the whole cyclic buffer is enqueued at
once) and modifying users of Samsung DMA wrappers in cyclic mode to
behave appropriately.
Signed-off-by: NTomasz Figa <tomasz.figa@gmail.com>
Acked-by: NLinus Walleij <linus.walleij@linaro.org>
Signed-off-by: NMark Brown <broonie@linaro.org>

Driver core clears the driver data to NULL after device_release or on probe
failure, so just remove it from here.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>