This patch adds a USB quirk for the Yamaha THR10C amp.
Signed-off-by: NTrulan Martin <trulanm@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a USB quirk for the Yamaha THR5A amp.
Signed-off-by: NTrulan Martin <trulanm@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a USB quirk for the Yamaha THR10 amp.
Signed-off-by: NTrulan Martin <trulanm@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:

  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
  ....

It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card->probing = 1.

Since the call itself during card->probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.
Reported-and-tested-by: NDaniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.

There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.

When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.

Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: NTorstein Hegge <hegge@resisty.net>
Reported-and-tested-by: NYves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix english in sound/drivers/Kconfig.
Signed-off-by: NPavel Machek <pavel@ucw.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models.  This patch revives them again.

Bugzilla: https://bugs.archlinux.org/task/34865Reported-and-tested-by: NTobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

These are being reported as being so noisy at high mic boost levels,
so they are unusable in practice.
Therefore artificially limit the boosts.

BugLink: https://bugs.launchpad.net/bugs/1089795Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Set the timeout for USB control set messages according to the USB 2
spec, using the macros from include/linux/usb.h.
The get timout becomes 5000 ms even though it is 500 ms in the
spec. This patch is required to run the Hercules RMX2 which needs a
timeout of 1240 ms.

More notes from author:
I still distinguish between set and get but as long both are 5000 ms
GCC will remove it anyway. IMHO this is more easy read and there is no
need to explain why we use a get timeout for set messages.
Signed-off-by: NDaniel Schürmann <daschuer@mixxx.org>
Acked-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Like the previous patch by Dan, we should clear the data to be
returned from certain compress ioctls, namely,
snd_compr_get_codec_caps() and snd_compr_get_params().
This time, we can simply replace kmalloc() with kzalloc().
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If the ->get_caps() function doesn't clear the buffer then there would
stack information leaked to userspace.  For example,
soc_compr_get_caps() can return success without clearing the buffer.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.

Only tested as far as build.
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Only tested as far as build.
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Previously we just hard coded all streams as playback streams, this
patch checks the DAI to see if it is a capture or playback stream. It is
worth noting that at this time only unidirectional streams are
supported.
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The app_pointer is managed locally by the compress core for memory
mapped DSPs but for DSPs that are not memory mapped this would have to
be manually updated from within the DSP driver itself, which is hardly
very idiomatic.

This patch switches to using the cumulative values to calculate the
available buffer space because these are already gracefully passed out
of the DSP driver to the compress core and otherwise should be
functionally equivalent.
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NRichard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: More updates for v3.10

The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window.  These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.

Use the generic dmaengine PCM driver instead of a custom implemention.  There is
a minor functional change, the ux500 PCM driver did not preallocate the audio
buffer, while the generic dmaengine PCM driver will do this.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Acked-by: NLee Jones <lee.jones@linaro.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.

The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.

ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.

This patch adds support for this by adding a boolean flag to the
audio format struct.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.

The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.

To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.

The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital

DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.

The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).

DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:

                                                  configured hardware
        176.4KHz   352.8kHz   705.6KHz     <----       sample rate

8-bit                2.8MHz     5.6MHz
16-bit    2.8Mhz     5.6MHz    11.2MHz

         `-----------------------------'
             actual DSD sample rates
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When pin default configs are overridden via patch option, these are
evaluated before fixups are applied.  Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.

We may reorder the execution, but an easier fix is simply to disable
this sanity check.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.
Signed-off-by: NWei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The flag bus->shutdown implies that the control elements might have
been already destroyed.  When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.

This patch just adds a check of the flag in the caller side for
avoiding such a crash.

Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily.  So let it be for a while.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The hardware revision of the codec is based at 0x40.  Subtract that
before convering to ASCII.  The same as it is done for 98095.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.

This patch is needed for playback and capturing on a codec-less systems,
as otherwise the PCM device nodes are not even created.
Signed-off-by: NStas Sergeev <stsp@users.sourceforge.net>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Tested-by: NShawn Guo <shawn.guo@linaro.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting imx to generic DMA engine PCM driver.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Tested-by: NShawn Guo <shawn.guo@linaro.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Unfortunately there are still quite a few platforms with a dmaengine driver
which do not support reporting the number of bytes left to transfer. If we want
to support these platforms in the generic dmaengine PCM driver we have.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Tested-by: NStephen Warren <swarren@nvidia.com>
Tested-by: NShawn Guo <shawn.guo@linaro.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Tested-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Linux 3.9-rc7