The following patch might introduce this call chain:
  PCM .pointer callback
  + fw_iso_context_flush_completions
    + packet callback
      + snd_pcm_period_elapsed
        + PCM .pointer callback
Recursive calls to the pointer callback are not possible due to the PCM
group locking, so avoid this by moving the period notification into
a separate tasklet.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>

> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized.  A short cut for booleans.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

In the PCM read/write loop, the driver calls snd_pcm_update_hw_ptr()
at each time at the beginning of the loop.  Russell King reported that
this hogs CPU significantly.

The current code assumes that the pointer callback is very fast and
cheap, also not too much fine grained.  It's not true in all cases.
When the pointer advances short samples while the read/write copy has
been performed, the driver updates the hw_ptr and gets avail > 0
again.  Then it tries to read/write these small chunks.  This repeats
until the avail really gets to zero.

For avoiding this situation, a simple workaround is to call
snd_pcm_update_hw_ptr() only once at starting the loop, assuming that
the read/write copy is performed fast enough.  If the available count
becomes short, it goes to snd_pcm_wait_avail() anyway, and this
processes right.
Tested-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When pdata->reset_pin is passed with a negative value (means gpio
is invalid), then chip->reset_pin will not be assigned to a vaule,
it will use default value 0. This will cause unexpected behavior.

So, add this patch to correct.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The upper limit of the available minors isn't necessarily 128 + unit,
but it's rather up to 256.  Fixing this allows more than 8 devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are left-over codes from the ancient days with the static device
number limitation of 8.  Actaully OSS can support up to 16 cards.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Since this is a generic API which should support any userspace interface
for reporting jacks update the documentation a little to make that a bit
clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add the PCI ID of the Asus Xonar DGX card; it's otherwise
identical with the DG.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

It contains non-standard call.
Reported-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Also be more specific about some details while at it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function

Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Takashi Iwai <tiwai@suse.de>

This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.

[fixed missing break by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This adds db gain information to M-Audio Fast Track Ultra (8R) devices.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().

[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Drop some struct members and definitions that became obsolete during
the refactorization of the driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames
are used in HDMI and DisplayPort to describe the parameters of the audio
stream. Hence, drivers for such devices may use these definitions to, for
instance, fill a CEA-861 data structure and pass it to a display driver
to configure an IP.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.

This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ep->fill_max is a 1 bit flag, thus it has to be boolean.
  sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
  sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion
Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound/usb/endpoint.c: In function ‘deactivate_urbs’:
sound/usb/endpoint.c:520:16: warning: unused variable ‘flags’ [-Wunused-variable]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Document the new streaming code and some of the functions so that
contributers can catch up easier.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This patch adds a new generic streaming logic for audio over USB.

It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.

A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.

With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.

In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This is needed for new card-wide list operations.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

release_firmware() deals gracefully with NULL pointers, no need to check first.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

At the point of this error-handling code, both regions and the dma have
been allocated, so free it as done in previous and subsequent
error-handling code.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

At the point of this error-handling code, HAVE_DSPCODEH may be undefined,
so free INITCODE and PERMCODE as done elsewhere.  A jump and label are
introduced to avoid code duplication.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: fixes for 3.4

A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.

name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.

remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1

really select MIC1, not Mixer out for recording

"Mixer out" for recording is not a "pin", adjust comment
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:

  $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
  nid = 0x10, verb = 0x300, param = 0xb126
  value = 0x0
  $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
  nid = 0x10, verb = 0xb00, param = 0x8001
  value = 0x0
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.

Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>