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* ALSA: hda - Kill the rest of snd_print*() usagesTakashi Iwai2014-06-251-1/+1
| | | | | | | Pass the codec object so that we can replace all the rest of snd_print*() usages with the proper device-specific print helpers. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - if statement not indentedDan Carpenter2014-05-141-1/+1
| | | | | | | The "break;" should be indented. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix registration of beep input deviceTakashi Iwai2014-02-281-1/+0
| | | | | | | | | | | | | | | | | | The beep input device is registered via input_register_device(), but this is called in snd_hda_attach_beep_device() where the sound devices aren't registered yet. This leads to the binding to non-existing object, thus results in failure. And, even if the binding worked (against the PCI object), it's still racy; the input device appears before the sound objects. For fixing this, register the input device properly at dev_register ops of the codec object it's bound with. Also, call snd_hda_detach_beep_device() at dev_disconnection so that it's detached at the right timing. As a bonus, since it's called in the codec's ops, we can get rid of the further call from the other codec drivers. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Replace with standard printkTakashi Iwai2014-02-251-21/+21
| | | | | | | | | | | | Use dev_err() and co for messages from HD-audio controller and codec drivers. The codec drivers are mostly bound with codec objects, so some helper macros, codec_err(), codec_info(), etc, are provided. They merely wrap the corresponding dev_xxx(). There are a few places still calling snd_printk() and its variants as they are called without the codec or device context. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2014-02-121-4/+4
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| * ALSA: hda - Fix inconsistent Mic mute LEDTakashi Iwai2014-02-071-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The current code for controlling mic mute LED in patch_sigmatel.c blindly assumes that there is a single capture switch. But, there can be multiple multiple ones, and each of them flips the state, ended up in an inconsistent state. For fixing this problem, this patch adds kcontrol to be passed to the hook function so that the callee can check which switch is being accessed. In stac_capture_led_hook(), the state is checked as a bitmask, and turns on the LED when all capture switches are off. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Make snd_hda_gen_spec_free() staticTakashi Iwai2014-02-101-2/+1
| | | | | | | | | | | | | | | | The last user of snd_hda_gen_spec_free() is patch_via.c, and we can rewrite it safely with snd_hda_gen_free(), so that snd_hda_gen_spec_free() can be a local function in hda_generic.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Move HDA_FIXUP_ACT_FREE call in snd_hda_gen_free()Takashi Iwai2014-02-101-0/+1
| | | | | | | | | | | | Now Realtek and Conexant codec parsers just call snd_hda_gen_free(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Avoid unnecessary verbs write in snd_hda_activate_path()Takashi Iwai2014-01-301-1/+1
|/ | | | | | | | ... by using snd_Hda_codec_update_cache() instead of *_write_cache(). Since all path elements should have been updated by this function, we are safe to assume that the cache contents are consistent. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Apply codec power_filter to FG nodesTakashi Iwai2014-01-131-2/+3
| | | | | | | Apply the codec->power_filter to the FG nodes in general for reducing hackish set_power_state ops override in patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Don't create duplicated ctls for loopback pathsTakashi Iwai2014-01-081-2/+4
| | | | | | | | | | | | | | | | AD1986A mic pins (0x1d and 0x1f) share the same widget for controlling the loopback volume/mute, but the generic parser didn't check it. This ended up with the duplicated controls for the same effect. This patch adds the check of the duplication for avoiding it. After this fix, there will be only one control although it affects both paths; this remaining issue should be fixed later in a different patch. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Kill EXPORT_SYMBOL_HDA()Takashi Iwai2013-12-191-22/+22
| | | | | | Replace all with the standard EXPORT_SYMBOL_GPL(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2013-12-161-1/+46
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| * ALSA: hda - Add static DAC/pin mapping for AD1986A codecTakashi Iwai2013-12-111-1/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | AD1986A codec is a pretty old codec and has really many hidden restrictions. One of such is that each DAC is dedicated to certain pin although there are possible connections. Currently, the generic parser tries to assign individual DACs as much as possible, and this lead to two bad situations: connections where the sound actually doesn't work, and connections conflicting other channels. We may fix this by trying to find the best connections more harder, but as of now, it's easier to give some hints for paired DAC/pin connections and honor them if available, since such a hint is needed only for specific codecs (right now only AD1986A, and there will be unlikely any others in future). Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Mute all aamix inputs as defaultTakashi Iwai2013-12-101-0/+24
| | | | | | | | | | | | | | | | | | | | Not all channels have been initialized, so far, especially when aamix NID itself doesn't have amps but its leaves have. This patch fixes these holes. Otherwise you might get unexpected loopback inputs, e.g. from surround channels. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Enable stereo mix as default for AD and VIA codecsTakashi Iwai2013-12-091-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | AD and VIA codecs had stereo mixer input enabled as default before moving to the generic parser, and people think the lack of such a regression. In this patch, the stereo mixer input is added back to the input selection if no auto-mic is available, and if it's not disabled explicitly via hint. This should satisfy most of demands, i.e. stereo mix on desktop machines like what it worked before, and it still keeps the new auto-mic feature on laptops. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Add missing initialization of aamix pathsTakashi Iwai2013-12-061-0/+18
| | | | | | | | | | | | | | | | The loopback mixing paths aren't initialized correctly at init callback. Mostly this is harmless as codecs usually set the mute state as default, but we still should make sure. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Allow capture-only configurationTakashi Iwai2013-12-061-1/+2
| | | | | | | | | | | | | | | | We have blindly assumed that all valid configurations should have either analog or digital playback, but there can be capture-only configurations. The parser shouldn't escape in such a case. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2013-11-281-21/+58
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| * ALSA: hda - Check leaf nodes to find aamix ampsTakashi Iwai2013-11-281-12/+45
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The current generic parser assumes blindly that the volume and mute amps are found in the aamix node itself. But on some codecs, typically Analog Devices ones, the aamix amps are separately implemented in each leaf node of the aamix node, and the current driver can't establish the correct amp controls. This is a regression compared with the previous static quirks. This patch extends the search for the amps to the leaf nodes for allowing the aamix controls again on such codecs. In this implementation, I didn't code to loop through the whole paths, since usually one depth should suffice, and we can't search too deeply, as it may result in the conflicting control assignments. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix hp-mic mode without VREF bitsTakashi Iwai2013-11-271-1/+1
| | | | | | | | | | | | | | | | | | When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN bit. Spotted during debugging old MacBook Airs. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Create Headhpone Mic Jack Mode when really neededTakashi Iwai2013-11-271-8/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When a headphone jack is configurable as input, the generic parser tries to make it retaskable as Headphone Mic. The switching can be done smoothly if Capture Source control exists (i.e. there is another input source). Or when user explicitly enables the creation of jack mode controls, "Headhpone Mic Jack Mode" will be created accordingly. However, if the headphone mic is the only input source, we have to create "Headphone Mic Jack Mode" control because there is no capture source selection. Otherwise, the generic parser assumes that the input is constantly enabled, thus the headphone is permanently set as input. This situation happens on the old MacBook Airs where no input is supported properly, for example. This patch fixes the problem: now "Headphone Mic Jack Mode" is created when such an input selection isn't possible. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Split the generic parser as an individual moduleTakashi Iwai2013-11-261-0/+4
|/ | | | | | | | Drop the hard dependency on the generic parser code and load / unload the generic parser code dynamically if built as a module. This allows us to avoid the generic parser if only HDMI/DP codecs are found. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Check keep_eapd_on before inv_eapdTakashi Iwai2013-11-121-2/+2
| | | | | | | | | | | | We don't change the EAPD bit in set_pin_eapd() if keep_eapd_on flag is set by the codec driver and enable is false. But, we also apply the flipping of enable value according to inv_eapd flag in the same function, and this confused the former check, handled as if it's turned ON. The inverted EAPD check must be applied after keep_eapd_on check, instead. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Introduce the bitmask for excluding output volumeTakashi Iwai2013-11-051-2/+6
| | | | | | | | | | | | | | | Add a bitmask to hda_gen_spec indicating NIDs to exclude from the possible volume controls. That is, when the bit is set, the NID corresponding to the bit won't be picked as an output volume control any longer. Basically this is just a band-aid for working around the issue found with CS4208 codec, where only the headphone pin has a volume AMP with different dB steps. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60811 Cc: <stable@vger.kernel.org> [v3.12+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add missing initial vmaster hook at build_controls callbackTakashi Iwai2013-10-251-1/+3
| | | | | | | | | | | | | | | The generic parser has a support of vmaster hook, but this is initialized only in the init callback with the check of the presence of the corresponding kctl. However, since kctl is NULL at the very first init callback that is called before build_controls callback, the vmaster hook sync is skipped there. Eventually this leads to the uninitialized state depending on the hook implementation. This patch adds a simple workaround, just calling the sync function explicitly at build_controls callback. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix inverted internal mic not indicated on some machinesDavid Henningsson2013-10-141-1/+1
| | | | | | | | | | | | | | | | | | | | The create_bind_cap_vol_ctl does not create any control indicating that an inverted dmic is present. Therefore, create multiple capture volumes in this scenario, so we always have some indication that the internal mic is inverted. This happens on the Lenovo Ideapad U310 as well as the Lenovo Yoga 13 (both are based on the CX20590 codec), but the fix is generic and could be needed for other codecs/machines too. Thanks to Szymon Acedański for the pointer and a draft patch. BugLink: https://bugs.launchpad.net/bugs/1239392 BugLink: https://bugs.launchpad.net/bugs/1227491 Reported-by: Szymon Acedański <accek@mimuw.edu.pl> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-v3.12' of ↵Takashi Iwai2013-08-231-3/+3
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.12 - DAPM is now mandatory for CODEC drivers in order to avoid the repeated regressions in the special cases for non-DAPM CODECs and make it easier to integrate with other components on boards. All existing drivers have had some level of DAPM support added. - A lot of cleanups in DAPM plus support for maintaining controls in a specific state while a DAPM widget all contributed by Lars-Peter Clausen. - Core helpers for bitbanged AC'97 reset from Markus Pargmann. - New drivers and support for Analog Devices ADAU1702 and ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson Microelectronics WM8997. - Support for building drivers that can support it cross-platform for compile test.
| * ALSA: hda - Fix missing mute controls for CX5051Takashi Iwai2013-08-121-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | We've added a fake mute control (setting the amp volume to zero) for CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but this feature was overlooked in the generic parser implementation. Now the driver lacks of mute controls on these codecs. The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE bits in each place checking the amp capabilities. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001 Cc: <stable@vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Mute the right widget in auto_mute_via_amp modeTakashi Iwai2013-08-131-6/+25
| | | | | | | | | | | | | | | | | | | | The current generic parser code assumes that always a pin widget controls the mute for an output blindly although it might be a different widget in the middle. Instead of the fixed assumption, check each parsed path and just pick up the right widget that has been already defined as a mute control. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Allow auto_mute_via_amp on bind mute controlsTakashi Iwai2013-08-131-3/+23
| | | | | | | | | | | | | | The auto-mute using the amp currently works only for a single amp on a pin. Make it working also with HDA_CTL_BIND_MUTE type, too. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Fix invalid multi-io creation on VAIO-Z laptopsTakashi Iwai2013-07-291-4/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | VAIO-Z laptops need to use the specific DAC for the speaker output by some unknown reason although the codec itself supports the flexible connection. So we implemented a workaround by a new flag, no_primary_hp, for assigning the speaker pin first. This worked until 3.8 kernel, but it got broken because the driver learned for a better multi-io pin mapping, and not it can assign two mic pins for multi-io. Since the multi-io requires to be the primary output, the hp and two mic pins are assigned in prior to the speaker in the end. Although the machine has two mic pins, one of them is used as a noise- canceling headphone, thus it's no real retaskable mic jack. Thus, at best, we can disable the multi-io assignment and make the parser behavior back to the state before the multi-io. This patch adds again a new flag, no_multi_io, to indicate that the device has no multi-io capability, and set it in the fixup for VAIO-Z. The no_multi_io flag itself can be used generically, added via a helper line, too. Reported-by: Tormen <my.nl.abos@gmail.com> Reported-by: Adam Williamson <awilliam@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Add snd_hda_jack_detect_state() helper functionTakashi Iwai2013-07-211-3/+5
|/ | | | | | | | | | | | | | | | | | | | | | | | snd_hda_jack_detect() function returns a boolean value for a jack plugged in or not, but it also returns always true when the corresponding pin is phantom (i.e. fixed). This is OK in most cases, but it makes the generic parser misbehaving about the auto-mute or auto-mic switching, e.g. when one of headphone pins is a fixed. Namely, the driver decides whether to mute the speaker or not, just depending on the headphone plug state: if one of the headphone jacks is seen as active, then the speaker is muted. Thus this will result always in the muted speaker output. So, the problem is the function returns a boolean, after all, although we need to think of "phantom" jack. Now a new function, snd_hda_jack_detect_state() is introduced to return these tristates. The generic parser uses this function for checking the headphone or mic jack states. Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for keeping compatibility in other driver codes. Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Replace the magic number 44 with constTakashi Iwai2013-06-281-6/+6
| | | | | | | | The char arrays with size 44 are for the name string of snd_ctl_elem_id. Define the constant and replace the raw numbers with it for clarifying better. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix the max length of control name in generic parserTakashi Iwai2013-06-281-1/+1
| | | | | | | | | | | | add_control_with_pfx() in hda_generic.c assumes a shorter name string for the control element, and this resulted in the truncation of the long but valid string like "Headphone Surround Switch" in the middle. This patch aligns the max size to the actual limit of snd_ctl_elem_id, 44. Cc: <stable@vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add auto_mute_via_amp flag to generic parserTakashi Iwai2013-06-251-1/+46
| | | | | | | | | | | | | | | | | | | Add a new flag, auto_mute_via_amp, to determine the behavior of the headphone / line-out auto-mute. When this flag is set, the generic driver mutes the speaker and line outputs via the amp mute of each pin, instead of changing the pin control values. This is introduced for devices that don't work expectedly with the pin control values; for example, some devices are known to keep enabling the speaker outputs no matter which pin control values are set on the speaker pins. The driver doesn't check actually whether the pins have the output amp caps, but assumes that the proper mixer (mute) controls are created on all these pins. If not the case, you can't use this flag for your device. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/via - Clean up duplicated codesTakashi Iwai2013-06-031-30/+12
| | | | | | | | The previous commit was written in the way to make the backport to 3.9.y easier, and left the duplicated open codes intentionally. Now let's clean up the duplicated codes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add keep_eapd_on flag to generic parserTakashi Iwai2013-06-031-0/+2
| | | | | | | | | | | VT1802 codec seems to reset EAPD of other pins in the hardware level, and this was another reason of the silent headphone output on some machines. As a workaround, introduce a new flag indicating to keep the EPAD on to the generic parser, and set it in patch_via.c. Reported-by: Alex Riesen <raa.lkml@gmail.com> Cc: <stable@vger.kernel.org> [v3.9] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Allow setting automute/automic hooks after parsingTakashi Iwai2013-06-031-9/+33
| | | | | | | | | | | | | | | | Some codec drivers (VIA codecs and some Realtek fixups) set the automute and automic hooks after calling snd_hda_gen_parse_auto_config(). In the current code, the hook pointers are referred only in snd_hda_gen_parse_auto_config() and passed to snd_hda_jack_detect_enable_callback(), thus changing the hook values won't change the actually called callbacks properly. This patch fixes this bug by setting the static functions as the primary callback functions for the jack detection, and let them calling the appropriate hooks dynamically. Cc: <stable@vger.kernel.org> [v3.9] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Check the activity of the NID to be powered downTakashi Iwai2013-05-161-2/+7
| | | | | | | | When an inactive path is powered down with spec->power_down_unused flag, we should check the activity of each widget in the path whether it's still referred from any active path. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Use the primary DAC for all aamix outputsTakashi Iwai2013-04-161-4/+7
| | | | | | | | | | When setting up the aamix output paths, use the primary DAC instead of the individual DAC for each output as default. Otherwise multiple DACs will be turned on for a single aamix widget, which results in doubly or more volumes, because the duplicated signals will be sent through all these DACs for a single stream. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix aamix activation with loopback control on VIA codecsTakashi Iwai2013-04-161-6/+16
| | | | | | | | | | | | | | | | | | | | | When we have a loopback mixer control, this should manage the state whether the output paths include the aamix or not. But the current code blindly initializes the output paths with aamix = true, thus the aamix is enabled unless the loopback mixer control is changed. Also, update_aamix_paths() called by the loopback mixer control put callback invokes snd_hda_activate_path() with aamix = true even for disabling the mixing. This leaves the aamix path even though the loopback control is turned off. This patch fixes these issues: - Introduced aamix_default() helper to indicate whether with_aamix is true or false as default - Fix the argument in update_aamix_paths() for disabling loopback Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Cc: <stable@vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-v3.10' of ↵Takashi Iwai2013-04-151-1/+1
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.10 A bunch of changes here, the most interesting one subsystem wise being Morimoto-san's work to create snd_soc_component which doesn't do much for now but will be pretty important going forwards: - Add a new component object type which will form the basis of moving to a more generic handling of SoC and off-SoC components, contributed by Kuninori Morimoto. - A fairly large set of cleanups for the dmaengine integration from Lars-Peter Clausen, starting to move towards being able to have a generic driver based on the library. - Performance optimisations to DAPM from Ryo Tsutsui. - Support for mixer control sharing in DAPM from Stephen Warren. - Multiplatform ARM cleanups from Arnd Bergmann. - New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
| * ALSA: hda/generic - fix uninitialized variableJiri Slaby2013-04-051-1/+1
| | | | | | | | | | | | | | | | | | changed is not initialized in path_power_down_sync, but it is expected to be false in case no change happened in the loop. So set it to false. Signed-off-by: Jiri Slaby <jslaby@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Handle Headphone Mic jack more genericDavid Henningsson2013-04-111-11/+0
| | | | | | | | | | | | | | | | Now that we have a flag for headphone mics, we can use that flag in the jack creation instead of creating the jack manually. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - allow "Headphone Mic" parser flagDavid Henningsson2013-04-111-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | This allows a specific mic to get the "Headphone Mic" name, in addition to the existing "Headset Mic" name. Also, it allows for a special mark: if the sequence number is set to 0xc, that's an indication to prefer it for headset mic, and if it's set to 0xd, that's an indication to prefer it for headphone mic. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Lower the badness for independent HP penaltyTakashi Iwai2013-03-221-1/+1
| | | | | | | | | | | | | | The lack of independent HP mode shouldn't be too bad, but currently its badness is set a bit too high. Let's lower it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Allow codec drivers to give own badness tablesTakashi Iwai2013-03-221-14/+12
| | | | | | | | | | | | | | | | | | | | | | The standard badness values don't seem to fit to all preferences. Some configuration prefer the side output over the headphone, some want the speaker over the surround, etc. This patch moves the badness table pointers into hda_gen_spec, so that the codec driver can override them. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'for-linus' into for-nextTakashi Iwai2013-03-221-0/+46
|\| | | | | | | | | | | | | | | | | | | | | | | | | Merge back for-linus branch for the badness table adjustment for VIA codecs * for-linus: ALSA: hda - Fix DAC assignment for independent HP ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader ALSA: hda - Fix typo in checking IEC958 emphasis bit ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls() ALSA: snd-usb: mixer: propagate errors up the call chain ALSA: usb: Parse UAC2 extension unit like for UAC1 ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
| * ALSA: hda - Fix DAC assignment for independent HPTakashi Iwai2013-03-211-0/+46
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang <LydiaWang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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