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Here's a patch that adds MIDI support through USB for one of the Access
Music synths, the VirusTI.
The synth uses standard USBMIDI protocol on its USB interface 3, although
it does signal "vendor specific" class. A magic string has to be sent on
interface 3 to enable the sending of MIDI from the synth (this string was
found by sniffing usb communication of the Windows driver). This is all
my patch does, and it works on my computer.
Please note that the synth can also do standard usb audio I/O on its
interfaces 2&3, which already works with the current snd-usb-audio driver,
except for the audio input from the synth. I'm going to work on it when I
have some time.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Extend the list of devices whose firmware does not expect more than one
USB MIDI packet in one USB packet.
bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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USB devices tends to represent dB ranges in different way than ALSA expects.
Add possibility to override these values and add guessed values for
SoundBlaster MP3+.
Also rename 'Capture Input Source' control to 'Capture Source' for
SoundBlaster MP3+ and Extigy.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The volume register is from 0..0x7f and 0..0x1a range is mute.
Also, fix mute combining in wm_vol_put(). The wrong behaviour was
noticed by Peter Christensen.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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After hours of debugging, I finally found the reason why some source
and runtime combination does not work. The PTP (page table pages)
address must be aligned. I am not sure how much, but alignment to
PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines
to ensure proper virtual -> physical address translation.
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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This is a cleanup for the dummy driver. The model kernel module parameter
is introduced to select the soundcard emulation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Allow TLV blocks that do not have any values; the smallest possible TLV
is an empty container or one where the information is only in the tag.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Creating a control with TLV_COMMAND access was not possible because
snd_ctl_new1() forgot to include it in the mask of allowable access
bits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The code in pcm_lib updating runtime->hw_ptr_interrupt expects
that runtime->boundary is divisible with runtime->period_size.
Thanks are going to Clemens Ladisch for the notice.
Fix the runtime->boundary calculation using buffer_size * period_size
as base and find a least common multiple for 32bit platforms when
the expression might overflow.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr
update functions" commit:
"It is possible for the status/delay ioctls to be called when the sound
card's pointer register alreay shows a position at the beginning of the
new period, but immediately before the interrupt is actually executed.
(This happens regularly on a SMP machine with mplayer.) When that
happens, the code thinks that the position must be at least one period
ahead of the current position and drops an entire buffer of data."
Return back the hw_ptr_interrupt variable. The last interrupt pointer
is always computed from the latest hw_ptr instead of tracking it
separately (in this case all hw_ptr checks and modifications might
influence also hw_ptr_interrupt and it is difficult to keep it
consistent).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate
pin to get captured samples instead zeros. Tested on Lenovo Thinkstation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O"
commit. New sleeping queue is introduced to separate user space and kernel
space wake_ups. runtime->nowake is renamed to twake (transfer wake).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Add experimental support for the Asus Xonar DS.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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As snd_seq_timer_set_tick_resolution() is always called with the same
three fields of struct snd_seq_timer, it suffices to give that as the
only parameter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The previous jiffies delta was 0 in all cases. Use hw_ptr variable to
store and print original value.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Add the STMicroelectronics ST7597 codec and an unknown codec
from the same manufacturer found on the Creative SB 128 card (CT4810).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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This model needs both 'Headphone Jack Sense' and 'Line Jack Sense' muted
for audible playback, so just add it to the ad1981 jack sense blacklist.
Cc: stable@kernel.org
Tested-by: Pete <x41215201@gmail.com>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The patch "sbawe: fix memory detection" fixed detection
for memoryless SB32 cards but broke detection of memory
above 512KB. This patch fixes the regression.
The patch has been tested on the SB32 card (CT3670) with
0MB, 2MB and 8MB memory installed.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6 into fixes
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The problem is that in the original code sound_nblocks could go up to 1024
which would be an array overflow.
This was found with a static checker and has been compile tested only.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The XO's audio hardware is wired up to allow DC sensors (e.g. light
sensors, thermistors, etc) to be plugged in through the microphone jack.
Add sound mixer controls to allow this mode to be enabled and tweaked.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The XO-1.5 has a microphone LED designed to indicate to the user when
something is being recorded.
This light is controlled by the microphone bias voltage and it is
currently coming on all the time.
This patch defers the microphone port configuration until when recording
is actually taking place, fixing the behaviour of the LED.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into devel
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Added patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea.
Signed-off-by: Ken Prox <kprox@users.sourceforge.net>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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The WM9705 and WM9703 ops are the same actually so use
the same code for both.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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When runtime->periods == 1 or when pointer crosses end of ring buffer,
the delta might be greater than buffer_size.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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As noted by pl bossart <bossart.nospam@gmail.com>, the PCM I/O routines
(snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls
until all samples are not processed.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them.
The main change is hw_ptr_interrupt variable removal to simplify code
logic. This variable can be computed directly from hw_ptr.
Ensure that updated hw_ptr is not lower than previous one (it was possible
with old code in some obscure situations when interrupt was delayed or
the lowlevel driver returns wrong ring buffer position value).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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In some debug cases, it might be usefull to see previous ring buffer
positions to determine position problems from the lowlevel drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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To increase code readability, convert send xrun_debug() argument to
use defines.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6
* 'for-2.6.33' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6:
ASoC: fixup oops in generic AC97 codec glue
ASoC: fix params_rate() macro use in several codecs
ASoC: fsi-ak4642: Remove ak4642_add_i2c_device
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Initialize the glue by calling snd_soc_new_ac97_codec() as is done
in other ASoC AC97 codecs. Fixes an oops caused by dereferencing
uninitialized members in snd_soc_new_pcms().
Run-tested on Au1250.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch converts the alc889 Aspire-specific powerdown to a generic
one. Like the previous effort, it currently only handles Front and PCM
but can be easily extended to cover other nids. The existing hook for
alc889 Aspire-specific remains enabled. Upon further testing, I've added
its use for ALC861_AUTO as well. Following patches will enable them for
other quirks.
Tested-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch ports powerdown fixes to AD198x. Currently we only turn off
Front and HP for suspend, but this is easily extended for additional
nids.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Rewrite the codes to use strict_strtoul() instead of simple_strtoul().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Check whether the given NID is a pin widget before storing the
user-defined pin configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Call snd_hda_shutup_pins() at suspend and free for avoiding click noises.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add a common helper function for clearing pin controls before suspend.
Use the pincfg array instead of looking through all widget tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent change for supporting dynamic beep device allocation caused
a problem resulting in Oops at reloading the driver. Also, it ignores
the error from input device registration.
This patch fixes the wrong check in snd_hda_detach_beep_device(), and
returns an error when the input device registration fails properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The beep control verbs don't need to be cached for resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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gpio_led, gpio_led_polarity and gpio_mute are added now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch fixes a build failure introduced by the patch
ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params [1]
by adding/moving the aaci struct to the right position.
The patch mentioned above merged common source parts into one function,
but unfortunately left out the aaci struct and consequently caused a
build failure e.g. for arm versatile_config [2]
References:
[1] http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=d3aee7996c30f928bbbbfd0994148e35d2e83084
[2] http://kisskb.ellerman.id.au/kisskb/buildresult/1893605/
Patch against Linus' tree.
Signed-off-by: Peter Huewe <peterhuewe@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use snd_hda_jack_detect() again for jack-sensing.
The triggering problem can be worked around with codec->no_trigger_sense
flag now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Analog Device codecs seem to have problems with the triggering of
pin-sensing although their pincaps give the trigger requirements.
Some reported that constant CPU load on HP laptops with AD codecs.
For avoiding this regression, add a flag to codec struct to notify
explicitly that the codec doesn't suppot the trigger at pin-sensing.
Tested-by: Maciej Rutecki <maciej.rutecki@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When we run the following commands in turn (with
CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0),
speaker-test -Dhw:0,3 -c2 -twav # HDMI
speaker-test -Dhw:0,0 -c2 -twav # Analog
The second command will produce sound in the analog lineout _as well as_
HDMI sink. The root cause is, device 0 "reuses" the same stream tag that
was used by device 3, and the "intelhdmi - sticky stream id" patch leaves
the HDMI codec in a functional state. So the HDMI codec happily accepts
the audio samples which reuse its stream tag.
The proposed solution is to remember the last device each azx_dev was
assigned to, and prefer to
1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used
2) or assign a never-used azx_dev for HDMI
With this patch and the above two speaker-test commands,
HDMI codec will use stream tag 8 and Analog codec will use 5.
The stream tag used by HDMI codec won't be reused by others, as long
as we don't run out of the 4 playback azx_dev's. The legacy Analog
codec will continue to use stream tag 5 because its device id is 0
(this is a bit tricky).
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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