- 08 5月, 2012 3 次提交
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由 Takashi Iwai 提交于
This reverts commit 785f857d. The commit causes a problem with the wrong D3 state after suspend because the call of hda_set_power_state() involves with the power-up sequence, which changes the power_count, and this confuses the resume sequence that checks the power_count as well. Originally, this go-to-D3 sequence should be a simple task without the power-up sequence. But, it'd need some proper sanity checks in the case of power-saved state, so it's not too easy to write now in the 3.4-rc cycle. In short, the safest option now is to revert this affecting commit. Of course, we need to clean up and robustify the power-saving code better for 3.5 kernel. Reported-by: NKonstantin Khlebnikov <khlebnikov@openvz.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The call for alc_auto_parse_customize_define() must be done after the fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work properly (e.g. HP RP5800 with ALC662 codec). Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Andre Schramm 提交于
snd_hdsp uses its own ioctls to acquire config- and status information. Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels. Signed-off-by: NAndre Schramm <andre.schramm@iosono-sound.com> Reviewed-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 07 5月, 2012 2 次提交
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由 Takashi Iwai 提交于
Reported-by: NPhilipp Matthias Hahn <pmhahn@pmhahn.de> Cc: <stable@kernel.org> [v3.3+] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Acer Aspire 5739G requires the same fix-up for 4930G to support the surround / bass speakers. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 06 5月, 2012 1 次提交
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由 Mark Hills 提交于
This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: NMark Hills <mark@pogo.org.uk> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 4月, 2012 1 次提交
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由 David Henningsson 提交于
According to the reporter, external mic starts to work if the laptop-dmic model is used. According to BIOS pin config, all pins are consistent with the alc269vb_laptop_dmic fixup, except for the external mic, which is not present. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/950490Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 4月, 2012 4 次提交
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由 Takashi Iwai 提交于
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the docking-station ports, but BIOS doesn't initialize for these pins. Thus, like the former X200, we need to set up the pins manually in the driver. The odd part is that the same PCI SSID is used for X200 and T400, thus we need to prepare individual fixup tables for cx5051 and others. Bugzilla entries: https://bugzilla.redhat.com/show_bug.cgi?id=808559 https://bugzilla.redhat.com/show_bug.cgi?id=806217 https://bugzilla.redhat.com/show_bug.cgi?id=810697Reported-by: NJosh Boyer <jwboyer@redhat.com> Reported-by: NJens Taprogge <jens.taprogge@taprogge.org> Tested-by: NJens Taprogge <jens.taprogge@taprogge.org> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Some output pins on Conexant chips have no HP control bit, but the auto-parser initializes these pins unconditionally with PIN_HP. Check the pin-capability and avoid the HP bit if not supported. Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
While refactoring the mute-LED handling for HP laptops, I messed up the polarity check in a wrong way. The red (or the mute-LED if any) should appear in the muted state, corresponding to GPIO on. Reported-by: NMikko Vinni <mmvinni@yahoo.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Through the transition to the auto-parser, the support for Quanta/Gericom KN1 got broken. There are two problems behind it: - This machine doesn't like the default COEF setup for ALC260 we take now as default - BIOS doesn't set the pins correctly at all; especially the machine uses only the pin 0x0f for both headphone and speaker This patch adds the fixup as a workaround for these issues. Reported-and-tested-by: NUros Vampl <mobile.leecher@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 4月, 2012 3 次提交
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由 Jesper Juhl 提交于
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the 'for (;;)' loop, if the 'badness' value returned from fill_and_eval_dacs() is negative, then we'll return from the function without freeing the memory we allocated for 'best_cfg', thus leaking. Fix the leak by kfree()'ing the memory when badness is negative. While I was there I also noticed some trailing whitespace in the function that I removed (along with all other trailing whitespace in the file) - it didn't seem worth-while to do that as two patches, so I hope it's OK that I just did it all as one patch. Signed-off-by: NJesper Juhl <jj@chaosbits.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Josh Boyer 提交于
A user reported that setting model=imac24 used to allow sound to work on their Mac Pro 5,1 machine. Commit 5671087f "Move ALC885 macpro and imac24 models to auto-parser" removed this model option. All Mac machines are now explicitly handled with a quirk and the auto-parser. This adds a quirk for the device found on the Mac Pro 5,1 machines. This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559 [sorted the new entry in the ID number order by tiwai] Reported-by: NGabriel Somlo <somlo@cmu.edu> Signed-off-by: NJosh Boyer <jwboyer@redhat.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
It's compatible with 8930G. Using the same fixup gives the proper 5.1 sound back. Reported-and-tested-by: NDany Martineau <dany.luc.martineau@gmail.com> Cc: <stable@kernel.org> [v3.3+] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 4月, 2012 1 次提交
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由 Takashi Iwai 提交于
Add GPIO1 setup explicitly for Acer Aspire 493x & co. This could be set by alc_auto_init_amp(), but it's safer to set it more explicitly in the fixup table. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 4月, 2012 1 次提交
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由 Takashi Iwai 提交于
Since there are still many Acer models that might not be covered by the current fixup table, let's add back a few typical model names so that user can test the fixup without recompiling. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 4月, 2012 1 次提交
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由 Fengguang Wu 提交于
Also remove two warnings when CONFIG_SND_DEBUG is not set: sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’: sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable] sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable] Signed-off-by: NWu Fengguang <fengguang.wu@intel.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 07 4月, 2012 6 次提交
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由 Michael Karcher 提交于
name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls affecting those pins. remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1 really select MIC1, not Mixer out for recording "Mixer out" for recording is not a "pin", adjust comment Signed-off-by: NMichael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Michael Karcher 提交于
CX20549 (ctx5045) doesn't accept data on index 1 for output pins, as shown in the following hda-var transaction: $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126 nid = 0x10, verb = 0x300, param = 0xb126 value = 0x0 $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001 nid = 0x10, verb = 0xb00, param = 0x8001 value = 0x0 Signed-off-by: NMichael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Michael Karcher 提交于
The ID used for detection of the BenQ R55E actually identifies the Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop series. Schematics on the internet clearly indicate that the "Port C" (analog input connected to record source #4 and mixer input #4) is unconnected. Playing an audio CD through analog playback (using cdplay from cdtools) produces no sound, even with the mixer input labelled "CD" enabled, and the volume control in the CD drive set to maximum. This indicates the connection is really not present. Signed-off-by: NMichael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Michael Karcher 提交于
The "input converter" widget of the CX20459 has only one input amplifier, expose that one as "Capture Volume/Capture Switch". The actual record source selection is already exposed through the separately installed input mux. Signed-off-by: NMichael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Michael Karcher 提交于
This includes renaming "Line In" to line, also in the mixer settings. Signed-off-by: NMichael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Michael Karcher 提交于
The CX20549 has only one single input amp on it's input converter widget. Fix printing of values in the codec file in /proc/asound. Signed-off-by: NMichael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 3月, 2012 1 次提交
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由 Eliot Blennerhassett 提交于
Make this function consistent with others in this module by returning 1 for error, instead of -ENOMEM (reverts function signature change from a938fb1e) Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 29 3月, 2012 2 次提交
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由 David Howells 提交于
Remove all #inclusions of asm/system.h preparatory to splitting and killing it. Performed with the following command: perl -p -i -e 's!^#\s*include\s*<asm/system[.]h>.*\n!!' `grep -Irl '^#\s*include\s*<asm/system[.]h>' *` Signed-off-by: NDavid Howells <dhowells@redhat.com>
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由 Dan Carpenter 提交于
This function returns zero or -ENOMEM, but because it's type is u16, the -ENOMEM gets changed to 65524. None of the callers care, but lets fix it anyway as a cleanup. Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 22 3月, 2012 1 次提交
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由 Takashi Iwai 提交于
The recent Realtek driver tries to assign an extra input via the headphone plug when only a single input source is found. The code worked on Samsung Q1, but it broke ASUS 1015 where the mic is a digital-mic and only a specific ADC works. This patch fixes the assignment of ADC in the shared mic/hp case. Instead of assuming the single ADC at first, reduce the ADCs after trying to assign both mic and HP pins. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42973 Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 3月, 2012 1 次提交
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由 Hans Verkuil 提交于
The tea575x-tuner module has been updated to use the latest V4L2 framework functionality. This also required changes in the drivers that rely on it. The tea575x changes are: - The drivers must provide a v4l2_device struct to the tea module. - The radio_nr module parameter must be part of the actual radio driver, and not of the tea module. - Changed the frequency range to the normal 76-108 MHz range instead of 50-150. - Add hardware frequency seek support. - Fix broken rxsubchans/audmode handling. - The application can now select between stereo and mono. - Support polling for control events. - Add V4L2 priority handling. And radio-sf16fmr2.c now uses the isa bus kernel framework. Signed-off-by: NHans Verkuil <hans.verkuil@cisco.com> Thanks-to: Ondrej Zary <linux@rainbow-software.org> Signed-off-by: NMauro Carvalho Chehab <mchehab@redhat.com>
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- 16 3月, 2012 2 次提交
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由 Takashi Iwai 提交于
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The irq handler must check whether the MPU401 instance is still alive. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 3月, 2012 2 次提交
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由 Takashi Iwai 提交于
Now the mute-LED is controlled without powersave hack, and the ifdefs must be removed. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Anssi Hannula 提交于
A previous commit af65cbf2 (ALSA: hdmi: fix printout of SAD sampling rates) fixed the sample rates shown in /proc/asound/cardX/eldY and kernel log to not be entirely wrong. However, a missing rate from the array added in the patch causes HDMI rates 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz to be shown as 96 kHz, 176.4 kHz, 192 kHz, and 384 kHz, respectively. Fix the reporting by adding the ALSA rate 64 kHz into the conversion array between 48 kHz and 88.2 kHz. Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: stable@kernel.org Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 3月, 2012 3 次提交
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由 Takashi Iwai 提交于
We need to resume two legacy registers to recover MIDI/FM functionality on S3/S4 resume, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Since it's not always safe to assume that the vmaster hook is purely the mute-LED control, add the flag indicating whether to expose the mute-LED enum control or not. Currently, conexant codec sets this off for non-HP laptops where EAPD may be used really as EAPD. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Create snd_hda_add_vmaster_hook() and snd_hda_sync_vmaster_hook() helper functions to handle the mute-LED in vmaster hook more commonly. In the former function, a new enum control "Mute-LED Mode" is added. This provides user to choose whether the mute-LED should be turned on/off explicitly or to follow the master-mute status. Reviewed-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 3月, 2012 4 次提交
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由 Takashi Iwai 提交于
Added the vmaster hook for controlling EAPD dynamically to Conexant auto-parser. When the Master is muted, EAPDs are turned off as well. This will fix the missing mute-LED control on some machines in addition to the more power-saving in the auto-parser mode. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
We've had ugly static handling of the mute-LED with a powersave hook for ALC269 HP laptops just like done in patch_sigmatel.c. This is now rewritten with the new vmaster hook and a fixup code. For that, the new fixup action, ALC_FIXUP_ACT_BUILD, is introduced. It's called after build_controls is called. The reason of this new action is that vmaster hook must be added at this stage (not in init or probe). Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The mute-LED is controlled in patch_sigmatel.c by (ab-)using the powersave hook. This can be now rewritten with the vmaster hook instead, which is much simpler and can work even without CONFIG_SND_HDA_POWER_SAVE kconfig. A drawback is that the mute-LED corresponds _only_ to the Master mixer switch instead of checking the whole DACs. But usually this shouldn't be a big problem as PA enables the mixer elements accordingly. Also, this patch changes the code to create vmaster always even on STAC9200 and STAC925x. The former "Master" on these chips are renamed as "PCM" now. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
It'll be used for adding hooks in later patches. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 3月, 2012 1 次提交
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由 Takashi Iwai 提交于
When the driver is changed to use vmaster or a new slave element is added by the improvement of the parser code, user may face often the silent output because of the muted slave mixer although Master volume is properly set. And they complain. And I get upset. Although such a mixer element should be initialized via "alsactl init", it'd be more user-friendly if the known output slaves are unmuted and set to 0dB so that user can control the volume only with Master as default. Since Master is still set muted as default even with this change, no risk of the speaker blow up, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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