- 28 11月, 2012 1 次提交
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由 Takashi Iwai 提交于
This is a preliminary patch for introducing a protection to access races of snd_array instances. Call snd_array_init() appropriately at the initialization time and don't call it twice. Also the allocations of codec-spec structs are cleaned up by helper functions in patch_sigmatel.c and patch_analog.c. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 11月, 2012 1 次提交
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由 Takashi Iwai 提交于
The free callback is called at the state where no extra verbs are executed, thus calling *_shutup() is useless, as it's checking the shutdown flag. Remove such superfluous calls. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 11月, 2012 1 次提交
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由 Takashi Iwai 提交于
Several bug reports suggest that the forcibly resetting IEC958 status bits is required for AD codecs to get the SPDIF output working properly after changing streams. Original fix credit to Javeed Shaikh. BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361Reported-by: NRobin Kreis <r.kreis@uni-bremen.de> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 9月, 2012 1 次提交
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由 David Henningsson 提交于
The standard name (and what PulseAudio picks up) is "Dock Mic", not "Docking Mic" or "Docking-Station". Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 8月, 2012 1 次提交
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由 Takashi Iwai 提交于
CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its behavior can be well controlled via the default value and module parameter. Let's just replace it with the standard CONFIG_PM. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 8月, 2012 1 次提交
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由 Takashi Iwai 提交于
Added a simple support of automute for the front HP jack to AD1882 stack model. Such an addition is basically an exception -- we really want to avoid the static quirk codes, but AD1882 parser isn't still ready for moving to the BIOS auto-parser yet. So, as a quick fix, I merged it for now. In near future, we really need the big clean up of patch_analog.c to move on to the auto-parser... Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 03 7月, 2012 1 次提交
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由 Takashi Iwai 提交于
Straightforward conversion to the new pm_ops from the legacy suspend/resume ops. Since we change vx222, vx_core and vxpocket have to be converted, too. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 09 5月, 2012 1 次提交
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由 Takashi Iwai 提交于
Just code shuffles. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 4月, 2012 2 次提交
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由 Takashi Iwai 提交于
Add a new helper function to guess the default VREF pin control bits for mic in. This can be used to set the pin control value safely matching with the actual pin capabilities. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
For setting the pin-control values more safely to match with the actual pin capability bits, a copule of new helper functions, snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are introduced. These are simple replacement of the codec verb write with AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out superfluous pin-control bits if they don't fit with the corresponding pin capabilities. Some codecs are screwed up or ignore the command when such a wrong bit is set. These helpers will avoid such secret errors. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 3月, 2012 1 次提交
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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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- 03 2月, 2012 1 次提交
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由 Takashi Iwai 提交于
In most cases, the slave strings for vmaster are identical between volumes and switches except for "xxx Volume" and "xxx Switch" suffix. Now snd_hda_add_vmaster() takes the optional suffix argument so that each string can be composed with the given suffix, and we can share the slave name strings in both volume and switch calls nicely. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 16 11月, 2011 1 次提交
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由 Takashi Iwai 提交于
Introduce a table containing the pins and their jack-detection states for avoiding the unnecessary verbs to check the pin status at each time. When the unsol event is enabled via snd_hda_jack_detect_enable(), it automatically adds the given NID to the table. Then the driver supposes that the codec driver will set the dirty flag appropariately when an unsolicited event is invoked for that pin. The behavior for reading other pins that aren't registered in the table doesn't change. Only the pins assigned to the table are cached, so far. In near futre, this table can be extended to use the central place for the unsolicited events of all pins, etc, and eventually include the jack-detect kcontrols that replace the current input-jack stuff. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 11月, 2011 1 次提交
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由 Paul Gortmaker 提交于
Lots of sound drivers were getting module.h via the implicit presence of it in <linux/device.h> but we are going to clean that up. So fix up those users now. Signed-off-by: NPaul Gortmaker <paul.gortmaker@windriver.com>
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- 23 9月, 2011 1 次提交
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由 Raymond Yau 提交于
- Add "AD198x Headphone" playback device for independent headphone playback while playing 7.1 surround using rear panel audio jacks. - Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using DAC0 instead of DAC1 (HDA_FRONT) was already added to all models. - Add "Independent HP" switch to enable/disable this playback device. When the switch is OFF, headphone use "copy front" mode to get the front channel as the green jack. When the switch is ON, you can play stereo sound through "AD198x Headphone" device to headphone while playing 7.1 surround sound through "AD198x Analog" device. The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog" is open. Signed-off-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 9月, 2011 1 次提交
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由 Raymond Yau 提交于
- use DAC0 instead of DAC1 for Port-A Headphone - assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp" Signed-off-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 7月, 2011 1 次提交
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由 Takashi Iwai 提交于
It makes little sense to enable power-saving without PM. This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM in all places. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 6月, 2011 1 次提交
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由 Takashi Iwai 提交于
Some HP laptops with AD1981 have SPDIF connections, but currently the driver disables it statically. Better to check the pin default config to judge whether to enable or disable the SPDIF. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 06 6月, 2011 1 次提交
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由 Stephen Warren 提交于
The SPDIF output controls apply to converter widgets. A future change will create a PCM device per pin widget, and hence a set of SPDIF output controls per pin widget, for certain HDMI codecs. To support this, we need the ability to virtualize the SPDIF output controls. Specifically: * Controls can be "unassigned" from real hardware when a converter is not used for the PCM the control was created for. * Control puts only write to hardware when they are assigned. * Controls can be "assigned" to real hardware when a converter is picked to support output for a particular PCM. * When a converter is assigned, the hardware is updated to the cached configuration. Signed-off-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 03 6月, 2011 2 次提交
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由 Raymond Yau 提交于
Check whether the pin supports EAPD in ad198x_power_eapd_write. Signed-off-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
In ad198x_power_eapd(), wrong pin NIDs are used for controlling EAPD for HP and Front outputs of AD1988/AD1989. These are actually same with the ones for AD1984 & co, port-A is 0x11 and port-D 0x12. Reported-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 5月, 2011 1 次提交
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由 Tony Vroon 提交于
The microphone input on the back panel (pink connector) stopped operating correctly after an upgrade from 2.6.35 to 2.6.38; the actual problem manifests itself as a lack of microphone bias voltage (VREF_HIZ) on node 0x17. With AD1988_6STACK_DIG the maximum bias voltage (VREF_80) is applied and the headset operates correctly. Signed-off-by: NTony Vroon <tony@linx.net> Tested-by: NDoug Redlich <pbrigade@nxltech.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 5月, 2011 1 次提交
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由 Adrian Wilkins 提交于
Compare pin type enum to the pin type and not the array index. Fixes bug#0005368. Signed-off-by: NAdrian Wilkins <adrian.wilkins@nhs.net> Cc: <stable@kernel.org> (2.6.37 and later) Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 02 5月, 2011 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
Also fixed the assignment of multiout.dac_nids to satisfy const. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 4月, 2011 1 次提交
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由 Takashi Iwai 提交于
EAPD power-down should be called also for normal shutup cases. Let's move to there. This also fixes the compile warnings when CONFIG_PM isn't set automatically. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 3月, 2011 1 次提交
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由 David Henningsson 提交于
For codec AD1984A, add a new model to support Dell Precision R5500 or the microphone jack won't work correctly. BugLink: http://bugs.launchpad.net/bugs/741516Tested-by: NKent Baxley <kent.baxley@canonical.com> Cc: stable@kernel.org Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 15 2月, 2011 1 次提交
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由 Raymond Yau 提交于
Signed-off-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 1月, 2011 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 Raymond Yau 提交于
Attached a patch which add a new model to support multi-streaming playback for ad1988. playback another stereo stream through the front panel headphone on device 2 while playback through the speakers connected to rear panel on device 0 at the same time. Tested with ad1988a rev2 codec on asus P5B-V motherboard. Signed-off-by: NRaymond Yau <superquad.vortex2@gmail.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 1月, 2011 1 次提交
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由 David Henningsson 提交于
BugLink: http://bugs.launchpad.net/bugs/697240 If the "Volume" suffix is not given, alsa-lib gets confused and loses the dB information at the simple element level. Boosts generally affects both playback and capture, as they are applied early in the chain. Hence no "Playback" or "Capture" in the suffix. Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 10月, 2010 1 次提交
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由 Takashi Iwai 提交于
The sticky PCM stream assignment introduced in 2.6.36 kernel seems causing problems on AD codecs. At some time later, the streaming no longer works by unknown reason. A simple workaround is to disable sticky-assignment for these codecs. Tested-by: NVasily Khoruzhick <anarsoul@gmail.com> Cc: <stable@kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 9月, 2010 1 次提交
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由 Luke Yelavich 提交于
BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/643891 Set the Dell Latitude E6400 (1028:0233) SSID to use AD1984_DELL_DESKTOP Cc: stable@kernel.org Signed-off-by: NLuke Yelavich <luke.yelavich@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 09 9月, 2010 3 次提交
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由 Takashi Iwai 提交于
This patch improves the input-source label strings to be generated from the pin information instead of fixed strings per AUTO_PIN_* type. This gives more suitable labels, especially for mic and line-in pins. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
We can assign multiple pins to a single role now, let's reduce the redundant FRONT_MIC and FRONT_LINE. Also, autocfg->input_pins[] is no longer used, so this is removed as well. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Keep char array in the input_mux item itself instead of pointing to an external string. This is a preliminary work for improving the input-mux name based on the pin role. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 8月, 2010 1 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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- 03 8月, 2010 1 次提交
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由 Jerone Young 提交于
Just as with the X301. The X300 does not have a way to do SPDIF either. It does not have a dock connector, nor does it have the SPDIF through the headphone jack. This patch fixes it so X300 does not show SPDIF, since it cannot do it. To add all Lenovo Thinkpads had different codec subsytem IDs: X300: http://launchpadlibrarian.net/34862838/Card0.Codecs.codec.0.txtSigned-off-by: NJerone Young <jerone.young@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 4月, 2010 1 次提交
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由 Takashi Iwai 提交于
When HP laptop with AD1981 codec is suspended and the docking-station is connected before the resume, the outputs get confused, and wrongly routed still to the speaker. This is because of a change in 2.6.34-rc1 ea52bf26 ALSA: hda: Add powerdown for Analog Devices HDA codecs The problem was the added resume callback that doesn't consider the modified init hook. The fix is simply remove the resume callback here and make the resume normally. This doesn't change any behavior intended in the commit above (for shutting down the sound at suspend) but only fixes the resume. Reported-and-tested-by: NFrans Pop <elendil@planet.nl> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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