Commit Graph

298448 Commits

Author SHA1 Message Date
Takashi Iwai
e182534d4b ALSA: usb-audio - Call get_min_max_*() after determining the name string
get_min_max_with_quirks() must be called after the control id name
string is determined, but the current code changes the id name string
after calling the function.

Reported-by: Christian Melki <christian.melki@ericsson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-15 08:35:00 +02:00
Jaroslav Kysela
f3af90517d ALSA: hda - add probe_mask=0x101 automatically for WinFast VP200 H
This patch just sets the codec probe_mask=0x101 value for the WinFast VP200 H
PCoIP card based on Teradici hardware matching the PCI subsystem vendor/device
IDs 3a21:040d. The user reported no codec detection issues without this
explicit codec configuration.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-15 08:34:48 +02:00
Jaroslav Kysela
b012513c66 ALSA: snd-aloop - improve the sample copy accurracy
Maintain both streams (playback, capture) synchronized. Previous code
didn't take in account the small byte count drifts caused by the irq
position rounding.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-15 08:34:38 +02:00
Clemens Ladisch
92b862c7d6 ALSA: firewire-lib: optimize packet flushing
Trying to flush completed packets is pointless when the pointer
callback was called from the packet completion callback; avoid it.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-14 10:43:43 +02:00
Clemens Ladisch
e9148dddc3 ALSA: firewire-lib: flush completed packets when reading PCM position
By flushing all completed but not yet reported packets before reading
the PCM hardware position, the granularity of the pointer is improved
from the interrupt interval to the packet size.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-14 10:43:36 +02:00
Clemens Ladisch
76fb878948 ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call
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>
2012-05-14 10:43:30 +02:00
Mark Hills
7df4a691fb ALSA: usb-audio: Fix comment
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>
2012-05-11 21:27:36 +02:00
Takashi Iwai
0910c216f7 ALSA: pcm - Optimize the call of snd_pcm_update_hw_ptr() in read/write loop
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>
2012-05-11 19:05:12 +02:00
Bo Shen
b2522f9262 ALSA: atmel/ac97c: correct the unexpected behavior when using uninitial value for reset pin
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>
2012-05-11 12:10:04 +02:00
Takashi Iwai
2abb80176c sound: allow the unit search until 256 in sound_core.c
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>
2012-05-08 17:27:03 +02:00
Takashi Iwai
779ae5a083 ALSA: Fix the card number limit of OSS-emulation
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>
2012-05-08 17:25:56 +02:00
Mark Brown
a2e888f0d7 ALSA: jack: Update documention to reflect other userspace interfaces
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>
2012-05-07 18:11:37 +02:00
Clemens Ladisch
76bc7a0d0a ALSA: oxygen: add Xonar DGX support
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>
2012-05-05 14:24:12 +02:00
Takashi Iwai
1a442cc3df ALSA: asihpi - Revert module_pci_driver conversion for asihpi.c
It contains non-standard call.

Reported-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-26 07:19:39 +02:00
Daniel Mack
07a5e9d4fd ALSA: snd-usb: fix some typos in endpoint.c documentation
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>
2012-04-24 20:16:18 +02:00
Takashi Iwai
e9f66d9b9c ALSA: pci: clean up using module_pci_driver()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 12:25:00 +02:00
Andrew Morton
68853fa30c ALSA: usb-audio: sound/usb/endpoint.c: suppress warning
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>
2012-04-24 08:10:10 +02:00
Takashi Iwai
baba2e0d2b ALSA: usb-audio: Add missing error checks in snd_ebox44_create_mixer()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:07:38 +02:00
Felix Homann
d34bf14851 ALSA: usb-audio: M-Audio Fast Track Ultra: Add effect controls
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>
2012-04-24 08:06:06 +02:00
Felix Homann
cfe8f97c82 ALSA: usb-audio: Rename Fast Track Ultra mixer quirk functions
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>
2012-04-24 08:02:11 +02:00
Felix Homann
25ee7ef8fa ALSA: usb-audio: Add TLV to M-Audio Fast Track Ultra controls
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>
2012-04-24 08:01:46 +02:00
Felix Homann
285de9c08b ALSA: usb-audio: Rename and export mixer_vol_tlv
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>
2012-04-24 08:01:27 +02:00
Felix Homann
8a4d1d397b ALSA: usb-audio: Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks.
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>
2012-04-24 08:00:45 +02:00
Daniel Mack
c89a5d9cac ALSA: snd-usb: remove refactorization left-overs
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>
2012-04-21 17:40:28 +02:00
Takashi Iwai
f6e94c372e Merge branch 'topic/cea861-audio' into topic/misc 2012-04-18 08:01:19 +02:00
Ricardo Neri
7ba1c40b53 ALSA: Add definitions for CEA-861 Audio InfoFrames
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>
2012-04-18 08:00:36 +02:00
Takashi Iwai
56599bb020 Merge branch 'topic/usb-endpoint' into topic/misc 2012-04-18 07:57:32 +02:00
Mark Hills
7536c301f8 ALSA: snd-usb-audio: Replace mixer for Electrix Ebox-44
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>
2012-04-15 15:40:08 +02:00
Mark Hills
284a8dd6f0 ALSA: snd-usb-audio: Skip un-parseable mixer units instead of erroring
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>
2012-04-15 15:39:55 +02:00
Takashi Iwai
22026c1a7b ALSA: usb: Remove obsoleted fields from struct snd_usb_substream
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>
2012-04-13 12:57:39 +02:00
Takashi Iwai
85f71932e5 ALSA: usb: Fix fill_max flag set
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>
2012-04-13 12:41:54 +02:00
Takashi Iwai
c5ee4ec828 ALSA: usb: Remove unused variable
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>
2012-04-13 10:27:28 +02:00
Daniel Mack
94c27215bc ALSA: snd-usb: add some documentation
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>
2012-04-13 10:25:24 +02:00
Daniel Mack
c75a8a7ae5 ALSA: snd-usb: add support for implicit feedback
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>
2012-04-13 10:24:32 +02:00
Daniel Mack
d399ff9593 ALSA: snd-usb: remove old streaming logic
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:23 +02:00
Daniel Mack
edcd3633e7 ALSA: snd-usb: switch over to new endpoint streaming logic
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>
2012-04-13 10:24:08 +02:00
Daniel Mack
8fdff6a319 ALSA: snd-usb: implement new endpoint streaming model
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>
2012-04-13 10:23:42 +02:00
Daniel Mack
596580d0ee ALSA: snd-usb: add snd_usb_audio-wide mutex
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>
2012-04-13 10:21:55 +02:00
Jesper Juhl
507230c999 ALSA: riptide: remove redundant NULL test before release_firmware()
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>
2012-04-10 08:42:33 +02:00
Julia Lawall
38be95dd3d ALSA: sound/isa/sscape.c: add missing resource-release code
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>
2012-04-10 08:42:07 +02:00
Julia Lawall
156d14da4c sound: sound/oss/msnd_pinnacle.c: add vfrees
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>
2012-04-10 08:41:54 +02:00
Takashi Iwai
c38f62b08d 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.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

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.
2012-04-07 12:28:00 +02:00
Michael Karcher
250f32747e ALSA: hda - clean up CX20549 test mixer setup
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>
2012-04-07 12:25:56 +02:00
Michael Karcher
51969d62c3 ALSA: hda - CX20549 doesn't need pin_amp_workaround.
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>
2012-04-07 12:25:47 +02:00
Michael Karcher
e6e03daecd ALSA: hda - Remove CD control from model=benq for CX20549
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>
2012-04-07 12:25:40 +02:00
Michael Karcher
cbf2d28e83 ALSA: hda - fix record volume controls of CX20459 ("Venice")
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:34 +02:00
Michael Karcher
3edbbb9ec5 ALSA: hda - Rename capture sources of CX20549 to match common conventions
This includes renaming "Line In" to line, also in the mixer settings.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:25 +02:00
Michael Karcher
4f32456e5e ALSA: hda - Fix proc output for ADC amp values of CX20549
The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-07 12:25:02 +02:00
Stephen Warren
8abe05c6eb ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed
the prototype of tegra_i2s_debug_add, but didn't update the dummy inline
used when !CONFIG_DEBUG_FS. Fix that.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # 3.3
2012-04-06 12:28:49 +01:00
Stephen Warren
3fec6b6d5a ASoC: set idle_bias_off=1 for all platform DAPM contexts
The ASoC core currently defaults to using STANDBY rather than OFF for
idle ASoC platform devices, which causes a permanent pm_runtime_get() on
them. This keeps the device active unnecessarily. This can be especially
problematic when the ASoC platform device and DAI device are the same
device.

The distinction between OFF and STANDBY is likely not relevant for ASoC
platform drivers, since they aren't analog devices. So, solve this issue
by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this
turns out to be a problem, this value could be sourced from the
snd_soc_platform_driver, similarly to soc_probe_codec().

Note: Prior to this change, this caused a large (10) runtime_active count
for the Tegra I2S controller even when not in use, and a leak in that
value as streams were started and stopped. This change probably hides a
bug.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-05 21:55:49 +01:00