Update the Xonar config texts with the latest information about the
Xonar DS, HDAV1.3 Slim, and Xense.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the select directive does not handle indirect dependencies, select
those explicitly in the driver sections.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the possibility to route a mix of the two channels of stereo data to
the center and LFE outputs. Due to a WM8766 restriction, all surround
and back channels also get the mixed L/R signal in this case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Automatically mute the speaker outputs as long as a headphone is plugged.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the polarity of the headphone detection pin is known, replace
the debugging message with a proper jack plug input device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the correct number, register bits, and names for the input switches.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the volume update latch bit was accidentally in the wrong
function, where it would prevent the MSB from being written, instead of
correctly ignoring it for cached values.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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>
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>
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>
Clean up the playback pointer callback function a bit, and make the
pointer check more strictly to avoid bogus pointers.
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>
Add a quirk to make the BOSS ME-25 work.
Many thanks to Kees van Veen.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the Roland/Cakewalk A-300PRO/A-500PRO/A-800PRO keyboards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the other logical device of the PCR-1 so that not only
the MIDI interface but also the audio interface works.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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>
The M-Audio Fast Track Ultra series devices did not play sound correctly
at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive
fixes this.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
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>
This device is similar to the M-Audio Delta 1010LT in that it uses the
AK4524VF ADC/DAC, but it does not use the CS8427 for SPDIF.
The SPDIF appears to be set up correctly, but I am not able to test it
as I do not have any devices that use it.
This patch makes the ADC/DAC's and the hardware mixer visible to apps
such as alsamixer and envy24control.
Signed-off-by: Garnet MacPhee <dhubsith@comcast.net>
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>
The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.
This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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
