The M-Audio Fast Track Ultra series devices did not play sound correctly
at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive
fixes this.
Signed-off-by: NFelix Homann <fexpop@web.de>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This device is similar to the M-Audio Delta 1010LT in that it uses the
AK4524VF ADC/DAC, but it does not use the CS8427 for SPDIF.

The SPDIF appears to be set up correctly, but I am not able to test it
as I do not have any devices that use it.

This patch makes the ADC/DAC's and the hardware mixer visible to apps
such as alsamixer and envy24control.
Signed-off-by: NGarnet MacPhee <dhubsith@comcast.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.

This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.

It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.

More information: Kernel bugzilla bug#16300

[A copmile warning fixed by tiwai]
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The detection and loading of firmeware on riptide driver has been broken
due to rewrite of some codes, checking the presense wrongly.
This patch fixes the logic again.

Reference: kernel bug 16596
	https://bugzilla.kernel.org/show_bug.cgi?id=16596

Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Gcc complains that ret might be used uninitialized:

sound/usb/format.c: In function ‘snd_usb_parse_audio_format’:
sound/usb/format.c:354: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:354: note: ‘ret’ was declared here
sound/usb/format.c:414: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:414: note: ‘ret’ was declared here

I suppose it could be uninitialized if there is ever a UAC_VERSION_3
released. Anyway this patch is worthwhile if only to silence the gcc
warning.
Signed-off-by: NDan Carpenter <error27@gmail.com>
Acked-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This is V2 of the patch, after feedback from Clemens and Daniel.

This patch adds SuperSpeed support to the USB drivers under sound/. It adds
tests for USB_SPEED_SUPER to the appropriate places that check for the USB
speed.

This patch has been tested with our SS USB3 device emulating a set of Yamaha
speakers and a Logitech microphone, but with the descriptors modified to add
USB3 support. It has also been tested with the real speakers and microphone,
to make sure that USB2 devices still work.
Signed-off-by: NPaul Zimmerman <paulz@synopsys.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Any subsequent revisions will have these configuration changes applied
by default.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Change the chip defaults to optimise performance of some of the DSP
functionality.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Its hardware is handled more fully by the new azt1605/azt2316 drivers.
Signed-off-by: NRene Herman <rene.herman@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This is a new driver for Aztech Sound Galaxy ISA soundcards based on the
AZT1605 and AZT2316 chipsets. It's constructed as two seperate drivers
for either chipset generated from the same source file, with (very)
minimal ifdeffery.

The drivers check the SB DSP version to decide if they are being loaded
for the right chip. AZT1605 returns 2.1 by default and AZT2316 3.1.
This isn't full-proof as the DSP version can actually be set through
software but it's close enough -- as far as I've been able to see, the
DSP version can not be stored in the EEPROM and the cards will therefore
startup with the defaults.

This distinction could (with the same success rate) also be used to
decide which chip we're looking at at runtime meaning a single, merged
driver is also an option but I feel it's actually nicer this way. A
merged driver would have to postpone translating the passed in resource
values to the card configuration until it knew which one it was looking
at and would need to postpone erring out on mpu_irq=10 for azt1605 and
mpu_irq=3 for azt2316.

The drivers have been tested on various cards. For snd-azt1605:

FCC-ID I38-MMSN811: Aztech Sound Galaxy Nova 16 Extra
FCC-ID I38-MMSN822: Aztech Sound Galaxy Pro 16 II

and for snd-azt2316:

FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB
FCC-ID I38-MMSN826: Trust Sound Expert DeLuxe Wave 32 (05201)
FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ (05202)
FCC-ID I38-MMSN837: Packard Bell ISA Soundcard 030069
FCC-ID I38-MMSN846: Trust Sound Expert DeLuxe 16-3D (06300)
FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D (06301)
FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D

826 and 846 were also marketed directly by Aztech and then known as:

FCC-ID I38-MMSN826: Aztech Sound Galaxy Waverider 32+
FCC-ID I38-MMSN846: Aztech Sound Galaxy Nova 16 Extra II-3D

Together, these cover the AZT1605 and AT2316A, AZT2316R and AZT2316-S
chipsets. All cards work fully -- full-duplex PCM, MIDI and FM. Full
duplex is a little flaky on some.

I38-MSN811 tends to not work in full-duplex but sometimes does with the
highest success rate being achieved when you first start the capture and
then a playback instead of the other way around (it's a CS4231-KL
codec).

The cards with an AD1845XP codec (my I38-MMSN826 and one of my
I38-MMSN830s) are also somewhat duplex-challenged. Sometimes full-duplex
works, sometimes not and this varies from try to try. This seems likely
to be a timing problem somewhere inside wss-lib.

I38-MMSN826 has an additional "ICS2115 WaveFront" wavetable synth
onboard that isn't supported yet. The wavetable synths on I38-MMSN847
and I38-MMSN852 are wired directly to the standard MPU-401 UART and the
AUX1 input on the codec and work without problem.

CD-ROM audio on the cards is routed to the codec "Line" input, Line-In
to its Aux input, and FM/Wavetable to its AUX1 input. I did not rename
the controls due to the capture source enumeration: I see that
capture-source overrides are hardcoded in wss-lib and this is just too
ugly to live.

Versus the old snd-sgalaxy driver these drivers add support for the
models without a configuration EEPROM (which are common), full-duplex,
MPU-401 UART and OPL3. In the future they might grow support for that
ICS2115 WaveFront synth on 826 and an hwdep interface to write to the
EEPROM on the models that have one.
Signed-off-by: NRene Herman <rene.herman@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The commit eb541337
    ALSA: hda - Make converter setups sticky
changes the semantics of snd_hda_codec_cleanup_stream() not to clean up
the stream at that moment but delay the action.  This broke the codes
expecting that the clean-up is done immediately, such as dynamic ADC
changes in some codec drivers.

This patch fixes the issue by introducing a lower helper,
__snd_hda_codec_cleanup_stream(), to allow the immediate clean up.
The original snd_hda_codec_cleanup_stream() is kept as is now.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When a device is plugged over HDMI, it passes some information in ELD
including the supported PCM parameters like formats, rates, channels.
This patch adds the check to PCM open callback of HDMI streams so that
only valid parameters the device supports are used.

When no device is plugged, the parameters the codec supports are used;
it's mostly all parameters the hardware can work.  This is for apps
that are started before device plugging and do probing (e.g. a sound
daemon), so that at least, probing would work even before the device
plugging.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAndrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This codec has been obsoleted by ADI, so add appropriate warnings to the
source tree to dissuade people from using in new designs based on driver
support.
Signed-off-by: NSonic Zhang <sonic.zhang@analog.com>
Signed-off-by: NMike Frysinger <vapier@gentoo.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NCliff Cai <cliff.cai@analog.com>
Signed-off-by: NMike Frysinger <vapier@gentoo.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

Added the entries for 92HD87B1/3 and 92HD87B2/4 codecs.
These are compatible with existing 83xxx codecs.
Signed-off-by: NCharles Chin <Charles.Chin@idt.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org

Two users report model=auto is needed to make the internal mic work properly.
BugLink: https://bugs.launchpad.net/bugs/495134Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

* Add missing codec IDs.
* Modify some existing codec names for discrete GPUs to match newly
  added IDs. Note: existing names were a mixture of marketing and
  engineering GPU names. Equally, there's no reason that codec IDs
  have to be specific to a particular GPU or board, so identify
  codecs in a less marketing-oriented fashion.
* Reformat codec ID table so it's easier to read, for me at least.
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The snd-aloop module allows redirecting of the PCM playback in the
kernel back to the user space using the standard ALSA PCM capture API.

The module also allows time synchronization with another timing source
and notifications of playback stream parameter changes.
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

The Conexant CX20584 with 141f:5068 seems compatible with other
cxt5066 code.  Just add the missing id.
Tested-by: NCristopher Camacho Leandro <ccamacho@linuxmail.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

  This provides a new model and pin config for the snd-hda-intel
92HD83XXX codec for hp laptop model dv7-4000, enabling the subwoofer.

Signed-off-by: Steven Eastland <seastland at gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

I discovered tonight that ALSA no longer sets up a stream for the second ADC
provided by the Realtek ALC260 HDA codec.  At some point alc_build_pcms()
started using stream_analog_alt_capture when constructing the second ADC
stream, but patch_alc260() was never updated accordingly.  I have no idea
when this regression occurred.  The trivial patch to patch_alc260() given
below fixes the problem as far as I can tell.  The patch is against 2.6.35.
Signed-off-by: NJonathan Woithe <jwoithe@physics.adelaide.edu.au>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Call the gpio reset platform function instead of using the flawed
ac97 functionality of the MPC5200(b)

From MPC5200B User's Manual:
"Some AC97 devices goes to a test mode, if the Sync line is high
during the Res line is low (reset phase). To avoid this behavior the
Sync line must be also forced to zero during the reset phase. To do
that, the pin muxing should switch to GPIO mode and the GPIO control
register should be used to control the output lines."
Signed-off-by: NEric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: NGrant Likely <grant.likely@secretlab.ca>

of_device is just an alias for platform_device, so remove it entirely.  Also
replace to_of_device() with to_platform_device() and update comment blocks.

This patch was initially generated from the following semantic patch, and then
edited by hand to pick up the bits that coccinelle didn't catch.

@@
@@
-struct of_device
+struct platform_device
Signed-off-by: NGrant Likely <grant.likely@secretlab.ca>
Reviewed-by: NDavid S. Miller <davem@davemloft.net>

So far, we reset the converter setups like the stream-tag, the
channel-id and format-id in prepare callbacks, and clear them in
cleanup callbacks.  This often causes a silence of the digital
receiver for a couple of seconds.

This patch tries to delay the converter setup changes as much as
possible.  The converter setups are cached and aren't reset as long
as the same values are used.  At suspend/resume, they are cleared
to be recovered properly, too.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Indent the branch of an if.

The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@r disable braces4@
position p1,p2;
statement S1,S2;
@@

(
if (...) { ... }
|
if (...) S1@p1 S2@p2
)

@script:python@
p1 << r.p1;
p2 << r.p2;
@@

if (p1[0].column == p2[0].column):
  cocci.print_main("branch",p1)
  cocci.print_secs("after",p2)
// </smpl>
Signed-off-by: NJulia Lawall <julia@diku.dk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAndrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NEric Bénard <eric@eukrea.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The NID 0x11 on HP dc5750 with ALC260 should be a speaker although BIOS
gives it as a line-out.  This patch adds a quirk to fix the pin config
so that the real line-out is used properly.

Reference: bnc#624118
	https://bugzilla.novell.com/show_bug.cgi?id=624118Signed-off-by: NTakashi Iwai <tiwai@suse.de>

My previous patch assumed that the DMA mode (represented by 3 lowest bits of
ALS4K_GCR99_DMA_EMULATION_CTRL register) is set to the default value 0. If
that's not the case, it might result in invalid mode to be set.
This patch fixes this potential problem.
Signed-off-by: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Enable burst mode to prevent dropouts during high PCI bus usage.
The card is useless in X without this because of dropouts when anything moves
on the screen (at least with PCI VGA card). Enabling this is also recommended
by the datasheet (page 48).
Signed-off-by: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In patch_alc269(), we initialize the primary capsrc so that the device
works from the beginning.  It issues CONNECT_SEL verb no matter which
widget is although some widget (e.g. 0x23) has no connection selection
but a mixer, which requires unmuting instead.

This patch fixes the initialization of capsrc by re-using the code as
a helper function.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

An Intel board needs a white-list entry to enable PC-beep.
Otherwise the driver misdetects (due to bogus BIOS info) and ignores
the PC-beep on 2.6.35.
Reported-and-tested-by: NLeandro Lucarella <luca@llucax.com.ar>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Since the pending periods are often bogus and take long time until
actually processed, it often results in a high CPU usage of the hd-audio
workq.  Overall it's better to have low CPU consumption by avoiding a
too tight loop rather than the wake-up timing accuracy.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Define constants for the HD-audio stream format bits, and replace the
magic numbers in codes.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix HDA beep frequency on IDT 92HD73xx and 92HD71Bxx codecs.
These codecs use the standard beep frequency calculation although the
datasheet says it's linear frequency.

Other IDT/STAC codecs might have the same problem.  They should be
fixed individually later.
Signed-off-by: NDaniel J Blueman <daniel.blueman@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>