Applications may want to read ELD information to
understand what codecs are supported on the HDMI
receiver and handle the a-v delay for better lip-sync.
ELD information is exposed in a device-specific
IFACE_PCM kcontrol. Tested both with amixer and
PulseAudio; with a corresponding patch passthrough modes
are enabled automagically.
ELD control size is set to zero in case of errors or
wrong configurations. No notifications are implemented
for now, it is expected that jack detection is used to
reconfigure the audio outputs.
ALSA: hda - Fix a regression of the position-buffer check
The commit a810364a0424c297242c6c66071a42f7675a5568
ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.
This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().
Dan Carpenter [Thu, 29 Sep 2011 06:10:48 +0000 (09:10 +0300)]
sound: oss: use strlcpy() in sound_timer_init()
sound_timer.info.name is a 32 character buffer. This function only
has one caller (in sound/oss/ad1848.c) and it passes as 128 character
buffer as "name". I don't know if this is a problem in real life,
and I doubt we're going to add more OSS drivers so it's unlikely to
become an issue. But we may as well take care of it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Allow patching with any vendor/subsystem ids
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Export the default mmap function, snd_pcm_lib_default_mmap().
The upcoming non-snooping support in HD-audio driver will use this
to override the mmap method.
ALSA: hda - Avoid unnecessary verbs to clear PCM formats
Since really_cleanup_stream() is called from both purity_inactive_streams()
and hda_cleanup_all_streams(), the verbs to clear the PCM channel and
format may be called multiple times unnecessarily.
This patch adds checks to skip these unneeded verbs.
ASoC: ssm2602: Support setting the oscillator and the clock output state
Currently the oscillator is always enabled and the clock output is always
disabled. This patch adds support for controlling the oscillator and clock
output state through snd_soc_dai_set_sysclk. Which makes it possible to
disable or enable them dynamically according to the requirements of the board
on which the CODEC is used.
This patch also slightly modifies the behavior as to when the oscillator is
going to be disabled in low-power states. Previously it would only be disabled
in BIAS_OFF, now it is also going to be disabled in BIAS_STANDBY, since no
components which depend on it should be active in this state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the the internal oscillator is powered down when entering BIAS_OFF
state, but not re-enabled when going back to BIAS_STANDBY. As a result the
CODEC will stop working after suspend if the internal oscillator is used to
generate the sysclock signal. This patch fixes it by clearing the appropriate
bit in the power down register when the CODEC is re-enabled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Clemens Ladisch [Mon, 26 Sep 2011 19:15:27 +0000 (21:15 +0200)]
ALSA: usb-audio: increase control transfer timeout
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers. Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.
The timeout parameter of snd_usb_ctl_msg() is now constant, so we can
drop it.
Reported-by: Felipe Balbi <balbi@ti.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Peter Ujfalusi [Mon, 26 Sep 2011 13:26:31 +0000 (16:26 +0300)]
ASoC: twl6040: No need to change delay during HF ramp
The Handsfree gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x3 raw, at 16 the gain
is -26dB.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 26 Sep 2011 13:26:30 +0000 (16:26 +0300)]
ASoC: twl6040: No need to change delay during HS ramp
The Headset gain have 2dB steps all the way, so there is no
reason to have different delays as we approaching to the
end of the scale.
The comment was also wrong, since we have 0dB at 0x0 raw at
one end of the range, and not in the middle.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 26 Sep 2011 13:26:25 +0000 (16:26 +0300)]
ASoC: twl6040: Combine the custom volsw get, and put functions
We can manage with one set of get, and put function for the gain
controls we need to handle with custom code due to the shadowing
of the register.
For both get, and put function we can call decide based on the
mc->rreg value, if we need to call the volsw, or the vlosw_2r
variant (in 2r case rreg is not 0).
Handling of the shadow values are the same for both type of
controls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 26 Sep 2011 13:05:58 +0000 (16:05 +0300)]
ASoC: omap-mcpdm: API to configure offset cancellation
The offset cancellation values can be different from board to board, even
on the same HW platform.
Provide a way for the machine drivers to configure the McPDM offset
cancellation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: hda/realtek - Don't detect LO jack when identical with HP
The spec->autocfg.line_out_pins[] may contain the same pins as hp_pins[]
depending on the configuration. When they are identical, detecting the
line_jack_present flag screws up the auto-mute because alc_line_automute()
is called unconditionally at initialization while it won't be triggered
by unsol events, thus the old line_jack_present flag is kept for the
whole run.
For fixing this buggy behavior, the driver needs to check whether the
line-outs are really individual, and skip if same as headphone jacks.
When the headphone pin is assigned as primary output to line_out_pins[],
the automatic HP-pin assignment by ASSID must be suppressed. Otherwise
a wrong pin might be assigned to the headphone and breaks the auto-mute.
Peter Ujfalusi [Mon, 26 Sep 2011 07:56:42 +0000 (10:56 +0300)]
ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’
Fixes the following compile time warning:
omap-mcbsp.c:519: warning: suggest explicit braces to avoid ambiguous ‘else’
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 21 Sep 2011 19:54:47 +0000 (20:54 +0100)]
ASoC: Add platform data for WM1250 EV1 GPIOs
The WM1250 EV1 has some GPIOs which can be used to control the behaviour
at runtime. Request them all if supplied and add a set_bias_level()
function to start and stop the clocks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 21 Sep 2011 20:29:47 +0000 (21:29 +0100)]
ASoC: Don't force bias on ground referenced devices
Currently we force all devices in the system to be at the same bias level.
This is due to concerns about power or pop/click impacts from either
ramping VMID or mismatching VMID on the analogue I/O lines between
connected devices but does mean we power devices up more often than we
really need to.
If a device flags idle_bias_off this will usually mean that it's either
all digital or ground referenced (in which case the idle and powered bias
levels are identical) so this concern does not apply and we can save some
power by leaving it off when not needed itself.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 23 Sep 2011 15:22:48 +0000 (16:22 +0100)]
ASoC: Convert WM8962 MICBIAS to a supply widget
A supply widget is generally clearer than a MICBIAS widget and a mic bias
is just a type of supply so use a supply widget for the MICBIAS. This also
avoids confusion with the routing when connected to multiple inputs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 23 Sep 2011 15:05:00 +0000 (16:05 +0100)]
ASoC: Support a wider range of sample rates on Speyside WM8962
As we've only got one audio interface and it is symmetric we can just set
SYSCLK based on the sample rate requested by the application layer. Provide
a default so bypass paths work before audio playback.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Thomas Pfaff [Thu, 22 Sep 2011 16:26:06 +0000 (18:26 +0200)]
ALSA: usb-audio - clear chip->probing on error exit
The Terratec Aureon 5.1 USB sound card support is broken since kernel
2.6.39.
2.6.39 introduced power management support for USB sound cards that added
a probing flag in struct snd_usb_audio.
During the probe of the card it gives following error message :
usb 7-2: new full speed USB device number 2 using uhci_hcd
cannot find UAC_HEADER
snd-usb-audio: probe of 7-2:1.3 failed with error -5
input: USB Audio as
/devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6
generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio]
on usb-0000:00:1d.1-2/input3
I can not comment about that "cannot find UAC_HEADER" error, but until
2.6.38 the card worked anyway.
With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl
stops in snd_usb_autoresume with -ENODEV.
Raymond Yau [Fri, 23 Sep 2011 11:03:25 +0000 (19:03 +0800)]
ALSA: HDA - Add Independent Headphone for all models of ad1988/ad1989
- Add "AD198x Headphone" playback device for independent headphone playback
while playing 7.1 surround using rear panel audio jacks.
- Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using
DAC0 instead of DAC1 (HDA_FRONT) was already added to all models.
- Add "Independent HP" switch to enable/disable this playback device.
When the switch is OFF, headphone use "copy front" mode to get the front
channel as the green jack.
When the switch is ON, you can play stereo sound through "AD198x Headphone"
device to headphone while playing 7.1 surround sound through "AD198x Analog"
device.
The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog"
is open.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Axel Lin [Fri, 23 Sep 2011 05:10:57 +0000 (13:10 +0800)]
ASoC: Remove unneeded mutex_init in wl1273_probe()
Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support",
snd_soc_register_codec() now does all the codec list and mutex init.
Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 23 Sep 2011 06:52:02 +0000 (09:52 +0300)]
ASoC: twl6040: No need to read the INTID register
Since our irq handler has been called, it is granted, that
the reason was either PLUGINT, or UNPLUGINT.
The INTID register has been checked in the MFD part of
twl6040 driver (twl6040-irq.c).
We have no reason to read from chip again here.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 23 Sep 2011 08:19:13 +0000 (11:19 +0300)]
ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
Attempt to change McBSP CLKS source while another stream is active is not
safe after commit d135865 ("OMAP: McBSP: implement functional clock
switching via clock framework") in 2.6.37.
CLKS parent clock switching using clock framework have to idle the McBSP
before switching and then activate it again. This short break can cause a
DMA transaction error to already running stream which halts and recovers
only by closing and restarting the stream.
This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce
l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a
severe timeout error and does BUG_ON().
Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk
if the McBSP is already active. This test should have been here just from
the beginning anyway.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Dan Carpenter [Fri, 23 Sep 2011 06:25:05 +0000 (09:25 +0300)]
ALSA: hdspm - cleanup __user tags in ioctl()
This makes the code cleaner and silences a Sparse complaint:
sound/pci/rme9652/hdspm.c:6341:23: warning: incorrect type in assignment (incompatible argument 4 (different address spaces))
sound/pci/rme9652/hdspm.c:6341:23: expected int ( *ioctl )( ... )
sound/pci/rme9652/hdspm.c:6341:23: got int ( static [toplevel] *<noident> )( ... )
sound/pci/rme9652/hdspm.c:6102:44: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6225:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6264:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6283:50: warning: dereference of noderef expression
sound/pci/rme9652/hdspm.c:6289:59: warning: dereference of noderef expression
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Wed, 21 Sep 2011 17:19:14 +0000 (18:19 +0100)]
ASoC: Add another DAPM stat for neighbour checks
The number of times we look at a potentially connected neighbour is just
as important as the number of times we actually recurse into looking at
that neighbour so also collect that statistic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:54 +0000 (11:05 +0300)]
ASoC/MFD: twl6040: Combine bit definitions for Headset control registers
Use one set of defines for the HS bits, since they are identical in both
control register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:53 +0000 (11:05 +0300)]
ASoC: twl6040/sdp4430: Change legacy DAI name
Change the legacy DAI name from "twl6040-hifi" to "twl6040-legacy" to
be more intuitive.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:52 +0000 (11:05 +0300)]
ASoC: twl6040: Support for AUX L/R output
AUX L/R outputs can be driver from the Handsfree PGA output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:51 +0000 (11:05 +0300)]
ASoC: twl6040: Use consistent names for Headset path
Use "Headset XYZ" for user visible controls, while the internal DAPM
widgets can use "HS XYZ".
In this way we can group the Headset related controls in UI
(alsamixer for example).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:50 +0000 (11:05 +0300)]
ASoC: twl6040: Use consistent names for Handsfree path
Use "Handsfree XYZ" for user visible controls, while the internal DAPM
widgets can use "HF XYZ".
In this way we can group the Handsfree related controls in UI
(alsamixer for example).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:49 +0000 (11:05 +0300)]
ASoC: twl6040: Earphone path correction
Fix the DAPM routing for the earphone path.
Convert the DAPM_SWITCH_E to DAPM_OUT_DRV_E, so we can have correct
power up, and down sequence for EP.
Introduce mute control (Earphone Playback Switch) for users to
enable/disable the EP path.
Note: the EP does not have it's own dedicated DAC. EP is connected to
HSL DAC.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:48 +0000 (11:05 +0300)]
ASoC: twl6040: Introduce SW only shadow register
Software only shadow register to be used by the driver.
For example Earpiece path will need this shadow register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:47 +0000 (11:05 +0300)]
ASoC: twl6040: Remove strings "NULL" from DAPM route
Replace the string with plain NULL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:46 +0000 (11:05 +0300)]
ASoC: twl6040: Fix comments for register names
Change the register name strings in the comments for the
twl6040_reg table, so it is easier to search for specific
register.
This is cosmetic change.
Before we had for example:
TWL6040_REG_HSLCTL as register definition.
At the register table we had:
TWL6040_HSLCTL
Searching for TWL6040_HSLCTL resulted no hits.
While if we look for REG_HSLCTL, we can find the places
the register has been used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Sep 2011 08:05:45 +0000 (11:05 +0300)]
ASoC: twl6040: Lower the power on gain values at startup
The default gains on outputs/inputs are set to 0dB.
This is fixing the pop noise issue at the first playback, which
caused by the wrong starting point of the ramp code.
The ramp code for the outputs expects the gains to be in
their lowest configuration in order to be effective.
After the playback stops, the ramp code takes care of
ramping down the gains to their minimum.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ben Hutchings [Thu, 22 Sep 2011 13:39:52 +0000 (14:39 +0100)]
ALSA: fm801: Gracefully handle failure of tuner auto-detect
Commit 9676001559fce06e37c7dc230ab275f605556176
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.
As a bonus, this should fix init and cleanup for the case where the
tuner is explicitly disabled.
Reported-and-tested-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946 Signed-off-by: Ben Hutchings <ben@decadent.org.uk> Cc: stable@kernel.org [v3.0+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
Once we have successfully called snd_device_new(), the cleanup
function fm801_free() will automatically be called by snd_card_free()
and we must *not* also call fm801_free() directly.
Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946 Signed-off-by: Ben Hutchings <ben@decadent.org.uk> Cc: stable@kernel.org [v3.0+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
projects / karo-tx-linux.git / log