Initialise model-specific DAC and ADC parts.
Add controls for output and mic source selection.
Rename some mixer controls according to ControlNames.txt.
Remove Playback switches for Line-in and IEC958-in - these
were controlling the input mute/unmute which affected
capture too. Use the capture switches to control the
input mute/unmute instead - it's less confusing.
Initialise the WM8775 to invert the left-right clock
to swap the left and right channels of the mic and aux
input.
Signed-off-by: Harry Butterworth <heb1001@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jesper Juhl [Mon, 13 Jun 2011 21:52:02 +0000 (23:52 +0200)]
ALSA: 6fire: Fix double-free bug in usb6fire_fw_ezusb_upload()
We have a double-free bug in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload().
We already call release_firmware(fw) on line 258, so when we then do it
again after usb6fire_fw_ezusb_write() returns <0, we have a double-free.
Easily fixed by just removing the last call to release_firmware().
Florian Zeitz [Sat, 11 Jun 2011 23:15:42 +0000 (01:15 +0200)]
ALSA: emu10k1: Add details for E-mu 0404 PCIe version
This patch adds the necessary details to support the PCIe version of
E-MU's 0404 card.
From comparing the PCBs it seems the PCIe version just added a PCIe
chipset and left all other components pretty much in place.
For anyone intrigued to take a look at the PCB there are pictures I took
at <http://babelmonkeys.de/~florob/E-MU%200404/>.
Adrian Knoth [Sun, 12 Jun 2011 15:26:18 +0000 (17:26 +0200)]
ALSA: hdspm - Fix jumping external wordclock frequency in AutoSync mode
When using Word Clock on RME MADI cards, AutoSync mode was alternating
betweeen MADI and WC due to a typo: AutoSync is indicated in the second
status register (status2), not the first one (status).
While the proc output was always correct, the reported WC frequency to
ALSA was unstable as mentioned in
Adrian Knoth [Sun, 12 Jun 2011 15:26:17 +0000 (17:26 +0200)]
ALSA: hdspm - Fix locking in snd_hdspm_midi_input_read
For the MIDI part, we need to acquire (and release) the hmidi->lock,
access to the global hdspm structure is serialized through
hmidi->hdspm->lock instead.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 10 Jun 2011 14:36:37 +0000 (16:36 +0200)]
ALSA: use KBUILD_MODNAME for request_irq argument in sound/pci/*
The name argument of request_irq() appears in /proc/interrupts, and
it's quite ugly when the name entry contains a space or special letters.
In general, it's simpler and more readable when the module name appears
there, so let's replace all entries with KBUILD_MODNAME.
BugLink: https://launchpad.net/bugs/761171
The original reporter needs the model=auto quirk for his internal
speakers to be audible in the latest daily snapshot, so add an entry in
the quirk table for his PCI SSID.
A trivially different version of this patch using the model=asus quirk
should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use
the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much
improved.
Reported-and-tested-by: tomdeering7 Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 10 Jun 2011 14:20:20 +0000 (16:20 +0200)]
ALSA: Use KBUILD_MODNAME for pci_driver.name entries
The convention for pci_driver.name entry in kernel drivers seem to be
the module name or equivalent ones. But, so far, almost all PCI sound
drivers use more verbose name like "ABC Xyz (12)", and these are fairly
confusing when appearing as a file name.
This patch converts the all pci_driver.name entries in sound/pci/* to
use KBUILD_MODNAME for more unified appearance.
Takashi Iwai [Fri, 10 Jun 2011 13:28:15 +0000 (15:28 +0200)]
ALSA: hda - Fix initialization of hp pins with master_mute in Realtek
Some Reatlek model quirks use master_mute bool switch for controlling
the master-mute of outputs. For these cases, the initialization of HP
pins/amps were forgotten during the transition to the common automute
helper function in 3.0 development time, and resulted in the muted HP
output as default.
This patch fixes the issue by adjusting the HP output explicitly with
master_mute switch.
Tested-by: Michal Hocko <mhocko@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 10 Jun 2011 12:56:26 +0000 (14:56 +0200)]
ALSA: hda - Fix SSYNC register value for non-Intel controllers
SSYNC register was once defined as 0x34-37 in the old Intel datasheet,
but corrected later to 0x38-3b. For fixing the register usage, a new
bit-flag is introduced for indicating the old ICH SSYNC register, and
ICH* PCI entries are added explicitly to enable this quirk.
Takashi Iwai [Fri, 10 Jun 2011 12:37:04 +0000 (14:37 +0200)]
ALSA: hda - Disable SPDIF only when no pin config set for HP with AD1981
Some HP laptops with AD1981 have SPDIF connections, but currently the
driver disables it statically. Better to check the pin default config
to judge whether to enable or disable the SPDIF.
Liam Girdwood [Thu, 9 Jun 2011 16:04:39 +0000 (17:04 +0100)]
ASoC: core - PCM mutex per rtd
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood [Thu, 9 Jun 2011 13:45:53 +0000 (14:45 +0100)]
ASoC: core - Separate out PCM operations into new file.
In preparation for Dynamic PCM support (AKA DSP support).
There will be future patches that add support to allow PCMs to be dynamically
routed to multiple DAIs at startup and also during stream runtime. This patch
moves the ASoC core PCM operaitions into a new file called soc-pcm.c. This will
in simplify the ASoC core features into distinct files.
Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context
Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a
DAPM context not matching the widgets context. This can lead to a wrong
prefix_len calculation, which will result in undefined behaviour. To avoid
this always use the DAPM context from the widget itself.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Timur Tabi [Wed, 8 Jun 2011 20:02:55 +0000 (15:02 -0500)]
ASoC: p1022ds: fix incorrect referencing of device tree properties
Device tree integer properties are encoded in big-endian format, but some of
the Freescale ASoC drivers were assuming that the host is in big-endian format
as well. Although this is true, it's better to use endian-safe accessors.
Also add a check for a failed ioremap() call in the SSI driver.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Wed, 8 Jun 2011 20:02:56 +0000 (15:02 -0500)]
ASoC: fsl: fix initialization of DMA buffers
The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs
for playback and capture, so DMA buffers should be allocated only for the
initialized streams. Instead of checking for the number of active channels,
which apparently is not reliable, check to see if the actual stream object
exists.
Also provide a better name for the DMA interrupt.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 7 Jun 2011 22:16:29 +0000 (23:16 +0100)]
ASoC: Report errors when we have a WM8962 IRQ and don't get FLL lock
We really should be getting the interrupt - if we don't get one it's very
likely that the configuration is incorrect and audio will fail. Also
increase the timeout substantially in this case for safety.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Fri, 3 Jun 2011 15:36:30 +0000 (16:36 +0100)]
ASoC: Suppress restore of default register values for rbtree cache sync
Currently the rbtree code will write out the entire register map when
doing a cache sync which is wasteful and will slow things down. Check
to see if the value we're about to write is the default and don't bother
restoring it if it is, either the value will have been retained or the
device will have been reset and holds the value already.
We should really store the defaults in the nodes but this resolves the
immediate issue.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Commit f97d0c6d5f94 ("ASoC: AD1836: Add input gain control for ADC2") contained
a typo in the register name, causing a build error. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Greg Dietsche [Mon, 6 Jun 2011 20:53:01 +0000 (15:53 -0500)]
ASoC: wm8940: remove unnecessary if statements
removing unnecessary if(ret) checks
This updated patch corrects a minor spelling problem in the commit message
and resolves two other (similar) issues found in wm8940.c by Jonathan Cameron.
Signed-off-by: Greg Dietsche <Gregory.Dietsche@cuw.edu> Acked-by: Jonathan Cameron <jic23@cam.ac.uk> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel T Chen [Mon, 6 Jun 2011 22:55:34 +0000 (18:55 -0400)]
ALSA: hda: Fix quirk for Dell Inspiron 910
BugLink: https://launchpad.net/bugs/792712
The original reporter states that sound from the internal speakers is
inaudible until using the model=auto quirk. This symptom is due to an
existing quirk mask for 0x102802b* that uses the model=dell quirk. To
limit the possible regressions, leave the existing quirk mask but add
a higher priority specific mask for the reporter's PCI SSID.
Reported-and-tested-by: rodni hipp Cc: <stable@kernel.org> [2.6.38+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: AD1836: Remove unused fields from private struct
The control_type field is never used, so it can be removed. The
control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: AD1836: Add AD1835/AD1837/AD1838/AD1839 support
The AD183X codec devices are mostly register compatible and can easily be
supported by the same driver. The main difference between those devices
is the number of DACs and ADCs.
This patch adjusts the driver to allocate the controls, DAPM widgets and
routes for the DACs and ADCs dynamically based on the chip type.
The AD1836 is a bit special in that it supports different modes for its second
ADC, so it needs some special handling. Right now the driver hardcodes the mode
to the differential PGA mode.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The different ADC and DAC controls follow the same scheme, so add some helper
macros for declaring them.
This should make the code a bit more readable and also decreases the code size
a bit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Mon, 6 Jun 2011 18:12:44 +0000 (19:12 +0100)]
ASoC: Manage Speyside system clocking only in bias management
Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 6 Jun 2011 18:13:23 +0000 (19:13 +0100)]
ASoC: Add context parameter to card DAPM callbacks
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Sat, 4 Jun 2011 10:34:43 +0000 (11:34 +0100)]
ASoC: Don't bring the CODEC up to full power for supplies and biases
If the only widgets active within a CODEC are supplies and micbiases we
are not passing audio, we are probably just doing microphone detection.
This will not generally require either fully accurate reference voltages
or much power so
If this turns out to be unsuitable for some systems we can provide a
facility to override this decision.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Sat, 4 Jun 2011 10:25:10 +0000 (11:25 +0100)]
ASoC: Specify target bias state directly as a bias state
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Allow more dynamic management of the device clocking by allowing BCLK to
be calculated when we set SYSCLK. This means that if the system is idle
when hw_params() runs then we don't try to use the SYSCLK used in that case
to set up the BCLK dividers, we can instead wait until a later point such
as bias level configuration. This makes it easier to manage low power modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Fri, 3 Jun 2011 16:49:46 +0000 (17:49 +0100)]
ASoC: Error out when FLL lock interrupt is not delivered on WM8915
When the FLL locks on the WM8915 an interrupt is generated. For safety
error out if we don't get that interrupt when the IRQ output of the
WM8915 is hooked up. Since we *really* expect an interrupt but the
threaded IRQ handler may take a bit longer than expected to get
scheduled also dramatically increase the delay in this case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
projects / karo-tx-linux.git / log