Commit 346a5c890 (OMAP: control: move plat-omap/control.h
to mach-omap2/control.h) in the linux-omap tree removed
plat/control.h and most of its callers. This one slipped
through - breaking the build as below when
CONFIG_SND_OMAP_SOC_MCPDM is defined. Fix this.
CC sound/soc/omap/omap-mcpdm.o
sound/soc/omap/omap-mcpdm.c:35: fatal error: plat/control.h: No such file or directory
compilation terminated.
make[3]: *** [sound/soc/omap/omap-mcpdm.o] Error 1
make[2]: *** [sound/soc/omap] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Anand Gadiyar <gadiyar@ti.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Paul Walmsley <paul@pwsan.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Paul Walmsley <paul@pwsan.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
Previously the OMAP McBSP ASoC driver implemented CLKS switching by
using omap_ctrl_{read,write}l() directly. This is against policy; the OMAP
System Control Module functions are not intended to be exported to drivers.
These symbols are no longer exported, so as a result, the OMAP McBSP ASoC
driver does not build as a module.
Resolve the CLKS clock changing portion of this problem by creating a
clock parent changing function that lives in
arch/arm/mach-omap2/mcbsp.c, and modify the ASoC driver to use it.
Due to the unfortunate way that McBSP support is implemented in ASoC
and the OMAP tree, this symbol must be exported for use by
sound/soc/omap/omap-mcbsp.c.
Going forward, the McBSP device driver should be moved from
arch/arm/*omap* into drivers/ or sound/soc/* and the CPU DAI driver
should be implemented as a platform_driver as many other ASoC CPU DAI
drivers are. These two steps should resolve many of the layering
problems, which will rapidly reappear during a McBSP hwmod/PM runtime
conversions.
Signed-off-by: Paul Walmsley <paul@pwsan.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.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>
The OMAP ASoC McBSP code implemented CLKR and FSR signal muxing via
direct System Control Module writes on OMAP2+. This required the
omap_ctrl_{read,write}l() functions to be exported, which is against
policy: the only code that should call those functions directly is
OMAP core code, not device drivers. omap_ctrl_{read,write}*() are no
longer exported, so the driver no longer builds as a module.
Fix the pinmuxing part of the problem by removing calls to
omap_ctrl_{read,write}l() from the OMAP ASoC McBSP code and
implementing signal muxing functions in arch/arm/mach-omap2/mcbsp.c.
Due to the unfortunate way that McBSP support is implemented in ASoC
and the OMAP tree, these symbols must be exported for use by
sound/soc/omap/omap-mcbsp.c.
Going forward, the McBSP device driver should be moved from
arch/arm/*omap* into drivers/ or sound/soc/*, and the CPU DAI driver
should be implemented as a platform_driver as many other ASoC CPU DAI
drivers are. These two steps should resolve many of the layering
problems, which will rapidly reappear during a McBSP hwmod/PM runtime
conversion.
Signed-off-by: Paul Walmsley <paul@pwsan.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.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>
The PCM proc files may open a race against substream close, which can
end up with an Oops. Use the open_mutex to protect for it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pm_qos_request isn't freed properly when OSS PCM emulation is used
because it skips snd_pcm_hw_free() call but directly releases the
stream. This resulted in Oops later.
Tested-by: Simon Kirby <sim@hostway.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/640254
In some cases a magic processing coefficient is needed to enable
the internal speaker on Dell M101z. According to Realtek, this
processing coefficient is only present on ALC269vb.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for laptop Toshiba Satellite C650D to have proper external HP and
external Mic support.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then
the "next device" should be -1. This function just returns device + 1.
But the main thing is that "device + 1" can lead to a (harmless) integer
overflow and that annoys static analysis tools.
[fix the case for device == SNDRV_RAWMIDI_DEVICE by tiwai]
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_parse_pin_def_config() has some workaround for re-assigning
some pins declared as headphones to line-outs. This didn't work properly
for some cases because it used memmove() stupidly wrongly.
Reference: Novell bnc#637263
https://bugzilla.novell.com/show_bug.cgi?id=637263
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error handling in snd_seq_oss_open() has several bad codes that
do dereferecing released pointers and double-free of kmalloc'ed data.
The object dp is release in free_devinfo() that is called via
private_free callback. The rest shouldn't touch this object any more.
The patch changes delete_port() to call kfree() in any case, and gets
rid of unnecessary calls of destructors in snd_seq_oss_open().
Fixes CVE-2010-3080.
Reported-and-tested-by: Tavis Ormandy <taviso@cmpxchg8b.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver doesn't probe the device properly because of left-over cfg[]
that isn't used at all for msnd-classic device. This is only for msnd-
pinnacle.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EeePC 1001HAG has a similar problem like other ASUS machine, which doesn't
set the codec SSID properly for indicating the beep capability.
To enable PC-beep again, put this to the whitelist.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the wrong "return" in the loop, a capture substream won't be
released at disconnection properly if the device is capture only and has
no playback substream. This caused Oops occasionally at the device
reconnection.
Reported-by: Kim Minhyoung <minhyoung.kim@lge.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Line and Mic inputs cannot be used at the same time, so the driver
has to automatically disable one of them if both are set. However, it
forgot to notify userspace about this change, so the mixer state would
be inconsistent. To fix this, check if the other control gets muted,
and send a notification event in this case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Nathan Schagen
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the WM8776 chip, this driver uses a different sample format and
more features than the Windows driver. When rebooting from Linux into
Windows, the latter driver does not reset the chip but assumes all its
registers have their default settings, so we get garbled sound or, if
the output happened to be muted before rebooting, no sound.
To make that driver happy, hook our driver's cleanup function into the
shutdown notifier and ensure that the chip gets reset.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Nathan Schagen
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Audio Class v2 support code in 2.6.35 added checks for the
bInterfaceProtocol field. However, there are devices (usually those
detected by vendor-specific quirks) that do not have one of the
predefined values in this field, which made the driver reject them.
To fix this regression, restore the old behaviour, i.e., assume that
a device with an unknown bInterfaceProtocol field (other than
UAC_VERSION_2) has more or less UAC-v1-compatible descriptors.
[compile warning fixes by tiwai]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For devices with more than one control interface, let's assume the first
one contains the audio controls. Unfortunately, there is no field in any
of the descriptors to tell us whether a control interface is for audio
or MIDI controls, so a better check is not easy to implement.
On a composite device with audio and MIDI functions, for example, the
code currently overwrites chip->ctrl_intf, causing operations on the
control interface to fail if they are issued after the device probe.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new model adds the following functionality to HP G60:
- Automute of internal speakers
- Autoswitch of internal/external mics
- Remove SPDIF not physically present
BugLink: http://launchpad.net/bugs/587388
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There were some new formats added in commit 15c0cee6c8 "ALSA: pcm:
Define G723 3-bit and 5-bit formats". That commit increased
SNDRV_PCM_FORMAT_LAST as well. My concern is that there are a couple
places which do:
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (dummy->pcm_hw.formats & (1ULL << i))
snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
}
I haven't tested these but it looks like if "i" were equal to
SNDRV_PCM_FORMAT_G723_24 or higher then we might read past the end of
the array.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The spinlock lock in sound_timer.c is used without initialization.
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If hw error is ignored, status is updated with invalid info.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I think this is a typo, debugfs_pop_time should not be executable.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimloogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The attached patch enables playback on a Sony VAIO machine.
BugLink: http://launchpad.net/bugs/618271
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In synchronous mode the SSI_SRCCR values are ignored. Instead
SSI_STCCR must be used for both receiving and transmitting.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds quirk for the Lenovo S10-3t so the headphone &
microphone jacks will now work.
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new sticky PCM parameter introduced the delayed clean-ups of
stream- and channel-id tags. In the current implementation, this check
(adding dirty flag) and actual clean-ups are done only for the codec
chip. However, with HD-audio architecture, multiple codecs can be
on a single bus, and the controller assign stream- and channel-ids in
the bus-wide.
In this patch, the stream-id and channel-id are checked over all codecs
connected to the corresponding bus. Together with it, the mutex is
moved to struct hda_bus, as this becomes also bus-wide.
Reported-and-tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Intel and Nvidia HDMI codec drivers have own implementations of
sticky PCM parameters. Now HD-audio core part already has it,
thus both setups conflict. The fix is simply remove the part in
patch_intelhdmi.c and patch_nvhdmi.c and simply call
snd_hda_codec_setup_stream() as usual.
Reported-and-tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/619439
This ThinkPad model needs External Amplifier muted for audible playback,
so set the inv_eapd quirk for it.
Reported-and-tested-by: Dennis Bell <dennis.bell@parkerg.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just added new codec ids. These are almost compatible with existing ones.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.
It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.
More information: Kernel bugzilla bug#16300
[A copmile warning fixed by tiwai]
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix capture mixer elements for ALC680 base model
- Support auto change ADC for recording from MIC
- Cancel capture source assigned in auto mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is not supported by current hardware revisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
The detection and loading of firmeware on riptide driver has been broken
due to rewrite of some codes, checking the presense wrongly.
This patch fixes the logic again.
Reference: kernel bug 16596
https://bugzilla.kernel.org/show_bug.cgi?id=16596
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: sound/usb/format: silence uninitialized variable warnings
MAINTAINERS: Add Ian Lartey as comaintaner for Wolfson devices
MAINTAINERS: Make Wolfson entry also cover CODEC drivers
ASoC: Only tweak WM8994 chip configuration on devices up to rev D
ASoC: Optimise DSP performance for WM8994
ALSA: hda - Fix dynamic ADC change working again
ALSA: hda - Restrict PCM parameters per ELD information over HDMI
sound: oss: sh_dac_audio.c removed duplicated #include
