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The codec_dai needs to be shutdown should the machine startup fails.
This patch adds another bailout tag for that case and rename the tag
for configuration failures.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch works around misbehaviour of Creative Creative VF0470 Live Cam
which reports 16 kHz sample rate for audio capture while actually producing
8 kHz stream.
Signed-off-by: Arseniy Lartsev <arseniy@fizlesh.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Conflicts:
sound/usb/usbaudio.c
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Remove some code that is no longer needed now that the relevant parts of
the driver have been tested.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar"
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/oss/coproc.h:7: ERROR: trailing whitespace
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible
sound/oss/v_midi.h:7: ERROR: trailing whitespace
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Norberto Lopes <nlopes.ml@gmail.com>
Acked-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In 2.6.33 ACL269 unsol event handler was changed to look up the pre-defined
pins, but the headphone pins aren't defined properly in each quirk.
This patch adds the missing definitions, and fixes the speaker auto-mute
regression on some ASUS (and possibly other) laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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Separate SH DMA headers into ones, commonly used by both drivers, and ones,
specific to each of them. This will make the future development of the
dmaengine driver easier.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
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Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
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Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
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Now most (if not all) PXA platforms have been switched to the new MFP
API, it's rather safe to remove these unnecessary pxa_gpio_mode() calls
in pxa2xx-ac97-lib.c now.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
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This is really pxa27x specific and should be kept in pxa27x.c. With this
newly introduced function, the original set_resetgpio_mode() is deprecated.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
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MFP registers are saved and restored by the mfp sys_device before all
other platform devices, and it is unnecessary here.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
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Conflicts:
arch/arm/plat-omap/Kconfig
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (252 commits)
ASoC: Check progress when reporting periods from i.MX FIQ handler
ASoC: Remove a unused variables from i.MX FIQ runtime data
ALSA: hda - Add/fix ALC269 FSC and Quanta models
ALSA: hda - Add ALC670 codec support
OMAP4: PMIC: Add support for twl6030 codec
ALSA: hda - remove unnecessary msleep on power state transitions
usb/gadget/{f_audio,gmidi}.c: follow recent changes in audio.h
ASoC: fsi: Modify over/under run error settlement
ASoC: OMAP4: Add McPDM platform driver
ASoC: OMAP4: Add support for McPDM
ASoC: OMAP: data_type and sync_mode configurable in audio dma
ALSA: hda - Add missing description in HD-Audio-Models.txt
ALSA: add support for Macbook Air 2,1 internal speaker
ALSA: usbaudio: consolidate header files
ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
ALSA: usbaudio: implement basic set of class v2.0 parser
ALSA: usbaudio: introduce new types for audio class v2
ALSA: usbaudio: parse USB descriptors with structs
ALSA: hda - enable snoop for Intel Cougar Point
ALSA: hda - Remove identical definitions for macmini3 model
...
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Add support for the Edirol UA-1000 to the UA-101 driver.
Both devices behave the same, so we just have to shuffle around some
interface numbers and name strings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Machine driver for DB1200 AC97 and I2S audio systems, intended as a proper
reference asoc machine for Alchemy-based systems. AC97/I2S can be selected
at boot time by setting switch S6.7.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Cc: Linux-MIPS <linux-mips@linux-mips.org>
Cc: alsa-devel@alsa-project.org
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
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DMA can only be done from physical addresses; move the "virt_to_phys"
source/destination buffer address translation from the dbdma queueing
functions (since the hardware can only DMA to/from physical addresses)
to their respective users.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
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Remove dbdma compat macros, move remaining users over to default
queueing functions and -flags.
(Queueing function signature has changed in order to give
a build failure instead of silent functional changes due
to the no longer implicitly specified DDMA_FLAGS_IE flag)
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
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* 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: (41 commits)
of: remove undefined request_OF_resource & release_OF_resource
of/sparc: Remove sparc-local declaration of allnodes and devtree_lock
of: move definition of of_chosen into common code.
of: remove unused extern reference to devtree_lock
of: put default string compare and #a/s-cell values into common header
of/flattree: Don't assume HAVE_LMB
of: protect linux/of.h with CONFIG_OF
proc_devtree: fix THIS_MODULE without module.h
of: Remove old and misplaced function declarations
of/flattree: Make the kernel accept ePAPR style phandle information
of/flattree: endian-convert members of boot_param_header
of: assume big-endian properties, adding conversions where necessary
of: use __be32 for cell value accessors
of/flattree: use OF_ROOT_NODE_{SIZE,ADDR}_CELLS DEFAULT for fdt parsing
of/flattree: use callback to setup initrd from /chosen
proc_devtree: include linux/of.h
of: make set_node_proc_entry private to proc_devtree.c
of: include linux/proc_fs.h
of/flattree: merge early_init_dt_scan_memory() common code
of: add 'of_' prefix to machine_is_compatible()
...
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git://opensource.wolfsonmicro.com/linux-2.6-asoc into topic/asoc
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Currently the i.MX FIQ handler is reporting periods as elapsed based
purely on a timer running in the CPU. This means that any clock
mismatch between the CPU and the audio subsystem can result in the
status reported to applications drifting away from the actual status
of the hardware. This is particularly likely at present since the
SSI driver is only capable of operating in slave mode so it's very
likely that the interface will be clocked from a different source.
Instead check the offset reported by the FIQ and only notify when we
have transferred at least one period, re-firing the timer if we didn't
do so. Also factor out the calculation of the timer expiry time for
make it a bit easier to experiment with.
Note that this only improves the situation, problems can still be
triggered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Specify proper quirk models for FSC and Quanta machines with ALC269 codec.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- Fixed alc_subsystem_id( ) typo and add new function.
- !(ass & 0x100000)) ==> Delete this check. It is unnecessary check.
- Add porti
- ALC670 support
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This will save ~15ms boot time.
The first 10ms sleep was introduced in commit d2595d86e5 for (buggy)
Cxt codecs, so better to limit the sleep to the problem hardware.
For the second 10ms sleep, the HDA spec says:
Power State[1:0]:
00: Node Power state (D0) is fully on.
01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog
playback) which must remain fully on.
10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state.
11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software
control. Note that any low power state set by software must retain sufficient operational capability to properly
respond to subsequent software Power State command.
So 10ms is actually the max wait time. It should be safe to
remove/reduce it and rely on the loop of 1ms-sleeps.
CC: Marc Boucher <marc@linuxant.com>
CC: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Zhang Rui <rui.zhang@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add ASoC interface for OMAP McBSP2 and McBSP3 sidetones.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
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In current FSI driver, playback function cares only overrun,
and capture function cares only underrun.
But playback function should had cared about underrun,
and capture function should had cared about overrun too.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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McPDM platform driver is configured to use sDMA in order to transfer
to/from memory. Support for interfacing with ABE will be added later.
McPDM dai currently supports up to 4 downlink channels and 2 uplink
channels simultaneously, as well as 88.2 and 96 KHz, and a sample
size of 32 bits.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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McPDM is the interface between Phoenix audio codec
and the OMAP4430 processor. It enables data to be transfered
to/from Phoenix at sample rates of 88.4 or 96 KHz.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Allow client drivers to set the data_type (16, 32) and the
sync_mode (element, packet, etc) of the audio dma transferences.
McBSP dai driver configures it for a data type of 16 bits and
element sync mode.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add support for Macbook Air 2,1 (late 2008) internal speaker and
headphones. Create a "mba21" model for snd-hda-intel.
Signed-off-by: Reimundo Heluani <rheluani@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.
Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.
Now things are also nicely prefixed which makes understanding the code
easier.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.
However, it allows using these devices for now, without mixer support.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:
* the number of streaming interfaces is now reported by an interface
association descriptor. The old approach using a proprietary
descriptor is deprecated.
* The number of channels per interface is now stored in the AS_GENERAL
descriptor (used to be part of the FORMAT_TYPE descriptor).
* The list of supported sample rates is no longer stored in a variable
length appendix of the format_type descriptor but is retrieved from
the device using a class specific GET_RANGE command.
* Supported sample formats are now reported as 32bit bitmap rather than
a fixed value. For now, this is worked around by choosing just one of
them.
* A devices needs to have at least one CLOCK_SOURCE descriptor which
denotes a clockID that is needed im the class request command.
* Many descriptors (format_type, ...) have changed their layout. Handle
this by casting the descriptors to the appropriate structs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch adds some definitions for audio class v2.
Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.
Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The channel mode definitions for macmini3 model are identical with mb5.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Check the card->codec on soc_resume to detect if the soc
device is properly initialized.
If the card->codec is NULL, than do not continue the resume
operation, since the device is not initialized properly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Add a headphones-only quirk for the Fujitsu Siemens D1289.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Marc Haber <mh+alsa201002@zugschlus.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Signed-off-by: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1].
Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE.
The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker.
$ lspci -vvnn | grep -A10 Audio
20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)
Subsystem: ASUSTeK Computer Inc. Device [1043:8290]
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 0, Cache Line Size: 64 bytes
Interrupt: pin A routed to IRQ 17
Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: HDA Intel
[1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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