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Now one can choose speaker configuration in e.g. PulseAudio mixer
Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch fixes the aut-mute setup on HP T5735 with ALC262 codec.
Instead of wrong amp, use pin control toggling for muting the speaker now.
Tested-by: Lee Trager <lee.trager@hp.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some codecs disable widgets used for output pins and reserve as vendor-
spec widgets. Thus we need to check the widget type and pin cap before
actually sending SET_EAPD verbs in the auto-configuration mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled
properly in alc268_new_analog_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the
recording doesn't work with model=auto because ALC262 parser sets the
wrong cap NIDs to choose the route and the default route for the sole
input pin wasn't initialized properly. This patch solves these issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type
"HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set
properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO)
either.
As per the documentation of find_mute_led_gpio(), these strings occur
in HP B-series systems - so, before scanning the SMBIOS strings, we need to
check if we're dealing with a B-series system.
Need to get confirmation from HP if this logic takes care of all the
systems. I'm trying to poke a friend there.
Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The capture-related mixer elements are missing with ALC861/ALC660 codecs
when quirks are present, due to missing call of set_capture_mixer().
Reference: Novell bnc#567340
http://bugzilla.novell.com/show_bug.cgi?id=567340
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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This patch adds support for automatically muting the speakers when headphones
are inserted, as well as relabelling the headphone widgets from the
non-standard "HP" to the standard "Headphone" for the mb5 model.
Signed-off-by: Alex Murray <murray.alex@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The alc664-mode4 model doesn't seem to fit with Toshiba NB205 correctly.
NB205 uses the pin 0x17 connected with the mixer 0x0f for the speaker
output, which isn't controlled by mode4 model at all.
Rather model=auto works fine as is on the latest driver, so let it back
again.
Tested-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The capture source or input source mixer element wasn't created properly
for ALC861-VD codec due to the wrong NID passed to
alc_auto_create_input_ctls().
References: Novell bnc#568305
http://bugzilla.novell.com/show_bug.cgi?id=568305
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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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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With `while (i++ < MAX_WRITE_RETRY)' i reaches MAX_WRITE_RETRY + 1 after the loop
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BugLink: https://bugs.launchpad.net/ubuntu/+bug/498863
This mainboard needs ac97_codec=0.
Cc: stable@kernel.org
Tested-by: Apoorv Parle <apparle@yahoo.co.in>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
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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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: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Postpone the mixer name setup after the codec patch since the codec
patch may change the codec name string in itself.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A machine with AMD CPU with Nvidia board doesn't work with MSI.
Reported-by: Robert J. King <peritus@gurunetwork.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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With the attached patch I am able to use the sound on a new IMac 27.
What works:
*) Internal speakers
*) Internal microphone
*) Headphone
I don't have an external mic or a SPDIF device to test the rest.
Signed-off-by: Rafael Avila de Espindola <rafael.espindola@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current Realtek code makes no specific provision for turning stuff
off. The codec chip is placed into low-power mode generically, but this
doesn't turn off any external hardware connected to it, in particular
external amplifiers.
This patch creates a hook function that is called by the codec
suspend/resume functions. It ought to disable any external hardware in a
device-specific way. I've implemented a generic ALC889 function that
sets the EAPD pin properly, and used it for the Acer Aspire 8930G which
can benefit from this feature.
On my laptop, this results in ~0.5W extra savings.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch removes some extra mixers that do nothing on the Acer Aspire
8930G.
The CD mixer is useless because the SATA DVD/Blu-Ray drive has no analog
audio output, and the Side mixer is useless because we max out at 6ch
anyway.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch just simplifies the 8930G verb array a bit. Just use the
common ALC889 EAPD verb array to make things more consistent. The file
is already huge enough already.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BugLink: https://bugs.launchpad.net/bugs/479373
The OR has verified with hda-verb that the internal microphone needs
VREF50 set for audible capture.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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 fixes
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Use kzalloc rather than kcalloc(1,...)
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
@@
- kcalloc(1,
+ kzalloc(
...)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Works fine with the auto-parser.
Reference: Novell bnc#564940
https://bugzilla.novell.com/show_bug.cgi?id=564940
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some model quirks missed the corresponding capsrc_nids. This resulted in
non-working capture source selection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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Conexant CX20583-10Z has digital beep device with volume control.
Making use of them.
Signed-off-by: Einar Rünkaru <einarry@smail.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixed initialization of internal mic and added internal mic boost control
Renamed analog mic boost control to ext mic boost contol.
Name pair analog/digital seems too confusing for a normal user.
Signed-off-by: Einar Rünkaru <einarry@smail.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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1. Add more ASUS NB model.
2. Fixed alc663_m51va_setup
M51VA has Digital Mic that NID is 0x12. The record source index is
0x9 for ALC663.
So, to modify the alc663_m51va_setup function to index 0x9
and add analog Mic aupport function alc663_mode1_setup.
3. Add ASUS mode7 and mode8 modules for ALC663
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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:
ALSA: ac97_codec - increase timeout for analog sections to 5 second
ASoC: Correct code taking the size of a pointer
ALSA: hda - Add PCI IDs for Nvidia G2xx-series
ALSA: sound/isa/gus: Correct code taking the size of a pointer
ALSA: hda: Fix max PCM level to 0 dB for AD1981_HP
ALSA: hda: Use ALC260_WILL quirk for another Acer model (0x1025007f)
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Makes use of skip_spaces() defined in lib/string.c for removing leading
spaces from strings all over the tree.
It decreases lib.a code size by 47 bytes and reuses the function tree-wide:
text data bss dec hex filename
64688 584 592 65864 10148 (TOTALS-BEFORE)
64641 584 592 65817 10119 (TOTALS-AFTER)
Also, while at it, if we see (*str && isspace(*str)), we can be sure to
remove the first condition (*str) as the second one (isspace(*str)) also
evaluates to 0 whenever *str == 0, making it redundant. In other words,
"a char equals zero is never a space".
Julia Lawall tried the semantic patch (http://coccinelle.lip6.fr) below,
and found occurrences of this pattern on 3 more files:
drivers/leds/led-class.c
drivers/leds/ledtrig-timer.c
drivers/video/output.c
@@
expression str;
@@
( // ignore skip_spaces cases
while (*str && isspace(*str)) { \(str++;\|++str;\) }
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- *str &&
isspace(*str)
)
Signed-off-by: André Goddard Rosa <andre.goddard@gmail.com>
Cc: Julia Lawall <julia@diku.dk>
Cc: Martin Schwidefsky <schwidefsky@de.ibm.com>
Cc: Jeff Dike <jdike@addtoit.com>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: "H. Peter Anvin" <hpa@zytor.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Cc: Neil Brown <neilb@suse.de>
Cc: Kyle McMartin <kyle@mcmartin.ca>
Cc: Henrique de Moraes Holschuh <hmh@hmh.eng.br>
Cc: David Howells <dhowells@redhat.com>
Cc: <linux-ext4@vger.kernel.org>
Cc: Samuel Ortiz <samuel@sortiz.org>
Cc: Patrick McHardy <kaber@trash.net>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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Previously, OLPC support for the mic extensions was only enabled in the
ALSA driver if CONFIG_OLPC and CONFIG_MGEODE_LX were both set. This was
because the old geode GPIO code was written in a manner that assumed
CONFIG_MGEODE_LX. With the new cs553x-gpio driver, this is no longer the
case; as such, we can drop the requirement on CONFIG_MGEODE_LX and instead
include a requirement on GPIOLIB.
We use the generic GPIO API rather than the cs553x-specific API.
Signed-off-by: Andres Salomon <dilinger@collabora.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jordan Crouse <jordan@cosmicpenguin.net>
Cc: David Brownell <david-b@pacbell.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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I have a Soundblaster 16PCI. For many years, alsa has had a bug where
not all of the card's controls are detected (many alsa versions,
many kernel versions). In particular, Master Playback Volume is
usually not detected, and so I get no sound or extremely faint sound.
The problem has always been inconsistent: sometimes all of the controls
are detected correctly, and sometimes a partial set is detected. It works
correctly about 10% of the time.
Finally, I got around to tracking down the problem. When the driver
fails, it prints the kernel message "AC'97 0 analog subsections not
ready". This message is generated from the function snd_ac97_mixer()
in ac97_codec.c. The message indicates that the card failed to come
back after reset within the time limit. The time limit is
120 milliseconds.
I tried increasing the time limit to 1 second, and found that this
made the driver work about 70% of the time. I tried increasing it
to 5 seconds, and it now seems to work 100% of the time.
I expect that this change would be completely harmless for
existing cards that work, and would only introduce additional
delay for cards that do not work.
ALSA bug#4032.
Signed-off-by: Steve Soule <sts11dbxr@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Signed-off-by: Stefan Ringel <stefan.ringel@arcor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BugLink: https://bugs.launchpad.net/bugs/461062
The original reporter states that PCM maxes at +12 dB and results in
very bad distortion. Cap PCM at 0 dB to resolve this symptom.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BugLink: https://bugs.launchpad.net/bugs/418627
The original reporter states that this quirk is necessary to obtain
reasonable gain for playback. Without it, sound is inaudible. Tested
with playback (spkr and hp) and capture.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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