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path: root/sound/pci/hda/hda_generic.c
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* ALSA: hda - Properly call automute/switch hooks at initTakashi Iwai2013-01-161-6/+23
| | | | | | ... and a little bit of code refactoring. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecsDavid Henningsson2013-01-161-2/+2
| | | | | | | Otherwise no PCM will be built for codecs without analog I/O. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - force different capture controls if amp caps differDavid Henningsson2013-01-161-3/+21
| | | | | | | | | Otherwise setting the capture volume for amps will be weird and inconsistent (it will try to set values outside the range of the second amp based on capabilities of the first amp). Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - do not add non-existing Mic boost controlsDavid Henningsson2013-01-161-0/+3
| | | | | | | | If the input node does not have any volume capable input amp, don't add such a control. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - initialize channel counts correctlyDavid Henningsson2013-01-161-1/+1
| | | | | | | | Even a single DAC can output two channels, so the channel count is twice the number of DACs. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - fix wrong adc_idx in generic parserDavid Henningsson2013-01-161-3/+3
| | | | | | | | We use knew->index for adc_idx when we create "Capture Volume" and "Capture Switch", so use the same to retrieve adc_idx. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Check array bounds in get_input_pathDavid Henningsson2013-01-161-0/+8
| | | | | | | This gives us some additional safety. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add prefer_hp_amp flag to hda_gen_specTakashi Iwai2013-01-151-4/+10
| | | | | | | | Add a new flag to indicate whether HP amp is turned on as default for speaker or line-outs, and enable this for ALC260 codec, as many machines with this codec require the HP amp even for speakers. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add capture_switch_hook to generic parserTakashi Iwai2013-01-141-1/+15
| | | | | | | Add a hook for the capture mixer switch. This will be used by IDT codecs for controlling the mic-mute LED. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Allow user to give hints for codec parser behaviorTakashi Iwai2013-01-121-0/+70
| | | | | | | Through the hints via sysfs or patch, user can set specific behavior flags for the generic parser now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add output jack mode enum controlsTakashi Iwai2013-01-121-0/+110
| | | | | | | | Add the enum controls for changing the headphone amp bits of output jacks, such as "Headphone Jack Mode". This feature isn't enabled as default, so far, unless spec->add_out_jack_modes flag is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Update automute / automic upon jack retaskingTakashi Iwai2013-01-121-0/+6
| | | | | | | | When a multi-io jack is switched to another direction, call the automute and autoswitch update functions, as this jack won't be used as the headphone or the mic jack that may turn off others. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Avoid auto-mute or auto-mic of retasked jacksTakashi Iwai2013-01-121-1/+8
| | | | | | | | | When a jack is retasked as a different direction (e.g. a mic jack is used as a CLFE output), such a jack shouldn't be counted as the target for the automatic jack switching. Skip the automute or the autoswitch when the current pinctl direction is different from what we suppose. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Manage current pinctl values in generic parserTakashi Iwai2013-01-121-52/+85
| | | | | | | | | | | | | | | Use the new pin target accessors for managing the current pinctl values in the generic parser. The pinctl values of all active pins are once determined at the initialization phase, and stored via snd_hda_codec_set_pin_target(). This will be referred again in the codec init or resume phase to set the actual pinctl. This value is kept while the auto-mute. When a line-out or a speaker pin is muted by auto-mute, the driver simply disables the pin, but it doesn't touch the cached pinctl target value. Upon unmute, this value is used to restore the original pinctl in return. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Clear the dropped paths properlyTakashi Iwai2013-01-121-1/+15
| | | | | | | | | | When a DAC is reassigned from surrounds to front or ADCs are reduced due to incomplete imux, we clear the path indices but the path instances remain as is. Since the paths might be still referred through the whole path list parsing (e.g. is_active_nid()), we should clear these path instances as well. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Allow aamix as a capture sourceTakashi Iwai2013-01-121-25/+44
| | | | | | | | | Since some codecs can choose the aamix as a capture source, we should support it as well. When spec->add_stereo_mix_input flag is set, the parser checks the availability of aamix as the input source, and adds the paths automatically when possible. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix inconsistent input_paths after ADC reductionTakashi Iwai2013-01-121-12/+22
| | | | | | | | | | | | In the current parser code, the input_paths[] may become inconsistent when some of detected ADCs are dropped due to incomplete inputs, since the driver rearranges only adc_nids[] but doesn't touch input_paths[]. This patch fixes the issue, and also it optimizes the reachability checks by simply referring to the parsed input_paths[] instead of calling is_reachable() again for each connection. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Return "Headphone Mic" from hda_get_autocfg_input_label()Takashi Iwai2013-01-121-4/+0
| | | | | | | | Instead of handling special cases in the caller side, give a proper name string "Headphone Mic" from hda_get_autocfg_input_label() when the headhpone jack pin is specified as an input. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Exclude aamix from capture pathsTakashi Iwai2013-01-121-6/+2
| | | | | | | The capture paths shouldn't contain the analog loopback mixer. Pass a proper argument to exclude the aamix NID. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add a flag to suppress mic auto-switchTakashi Iwai2013-01-121-0/+3
| | | | | | | Add a new flag spec->suppress_mic_auto_switch for codecs that don't support unsol events properly like VT1708. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Re-define snd_hda_parse_nid_path()Takashi Iwai2013-01-121-47/+55
| | | | | | | | | | | | This commit modifies the definition of snd_hda_parse_nid_path() slightly, now with_aa_mix argument is changed to anchor_nid, so that it can handle any NID generically as an anchor point to include or exclude. The with_aa_mix field in struct nid_path is removed again by this change. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Manage input paths via path indicesTakashi Iwai2013-01-121-26/+18
| | | | | | | ... like we did for output and loopback paths. It makes the code slightly easier. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix multi-io channel mode managementTakashi Iwai2013-01-121-30/+62
| | | | | | | | | | | The multi-io channels can vary not only from 1 to 6 but also may vary from 6 to 8 or such. At the same time, there are more speaker pins available than the primary output pins. So, we need three variables to check: the minimum channel counts for primary outputs, the current channel counts for primary outputs, and the minimum channel counts for all outputs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Remove unused dac reference in create_multi_out_ctls()Takashi Iwai2013-01-121-4/+0
| | | | | | Remove useless code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Use direct path reference in assign_out_path_ctls()Takashi Iwai2013-01-121-12/+16
| | | | | | | | Instead of looking through paths with the dac -> pin connection at each time, just pass the already parsed path index to assign_out_path_ctls(). This simplifies the code a bit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Clear path indices properly at each re-evaluationTakashi Iwai2013-01-121-0/+10
| | | | | | | | The path indices must be reset at each evaluation of DAC assignment. Otherwise the badness value will be wrongly calculated and mixers may be inconsistently assigned. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add brief comments to exported snd_hda_gen_*_() functionsTakashi Iwai2013-01-121-1/+12
| | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Remove dead HDA_CTL_BIND_VOL and HDA_CTL_BIND_SW codesTakashi Iwai2013-01-121-4/+0
| | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add snd_hda_gen_free() and snd_hda_gen_check_power_status()Takashi Iwai2013-01-121-15/+18
| | | | | | | Just to remove duplicated codes. Also fixed EXPORT_SYMBOL() in hda_generic.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add pcm_playback_hook to hda_gen_specTakashi Iwai2013-01-121-5/+60
| | | | | | | | The new hook which is called at each PCM playback ops. It can be used to control the codec-specific power-saving feature in each codec driver. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Drop bind-volume workaroundTakashi Iwai2013-01-121-97/+17
| | | | | | | | | | | | | | | | | | | | | | | | The bind-volume workaround was introduced for simplifying the mixer abstraction in the case where one or more pins of multiple outputs lack of individual volume controls. This was essentially the case like Acer Aspire 5935, which has 5.1 speakers and 5.1 (multi-io) jacks although there are 5 DACs, so some of them must share a DAC. However, the recent code rewrite changed the DAC assignment policy to share with the same channel instead of binding to the front, thus binding the volumes for all channels makes little sense now, rather it's confusing. So in this patch, the ugly workaround is finally dropped and simply create the volume control corresponding to the parsed path position. For dual headphones or 2.1 speakers with a shared volume control, it's anyway bound to the same DAC if needed, so this change shouldn't bring any practical difference. And, as a good bonus, we can cut off the whole code handling the bind volume elements. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Drop unneeded pin argument from set_output_and_unmute()Takashi Iwai2013-01-121-28/+13
| | | | | | Just a minor refactoring. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add missing slave names for Speaker Surround, etcTakashi Iwai2013-01-121-0/+3
| | | | Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Prefer binding the primary CLFE outputTakashi Iwai2013-01-121-6/+27
| | | | | | | | | | | When 5.1 or more multiple speakers with found but not enough DACs are available, it's better to bind such pins to the DACs of the primary outputs with the same channels rather than binding to the first DAC (i.e. the front channel). For the cases with two speaker pins, it's rather regarded as front + bass combination, thus it's more practical to still bind to the front, though. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix truncated control namesTakashi Iwai2013-01-121-2/+2
| | | | | | | | ... like "Speaker Surround Playback Switch". This fix had been already applied to patch_conexant.c but was forgotten in other places, then migrated to hda_generic.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add Loopback Mixing controlTakashi Iwai2013-01-121-0/+106
| | | | | | | | | | | | | | For codecs that have individual routes going through a loopback mixer (like VIA codecs), we need to provide an explicit switch to choose whether the output goes through mixer or directly from DAC. This patch adds the check for such paths and creates "Loopback Mixing" enum control when available. It won't influence on codecs like Realtek or others where the loopback mixer is connected independently from the primary output routes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Correct aamix output pathsTakashi Iwai2013-01-121-2/+2
| | | | | | | | | The output paths including aamix should be parsed only for the first output. The surround paths including aamix must be wrong, since it would mix all streams, i.e. all channels would be mixed into a single and multiplexed again. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Initialize digital-input path properlyTakashi Iwai2013-01-121-1/+7
| | | | | | | | Call the path activation for the digital input pin properly, not only setting the pin control. Also add spec->digin_path for keeping the path index. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Manage using output/loopback path indicesTakashi Iwai2013-01-121-50/+89
| | | | | | | | | | | | | | | Instead of search for the path with the certain route at each time, keep the path index for each output and loopback, and just use it when referred. In this implementation, the path index number begins with one, not zero (although I've been writing in C over decades). It's just to make the check for uninitialized values easier. So far, the input paths aren't handled with indices yet, but still picked up via snd_hda_get_nid_path() at each time. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix multi-io pin assignment in create_multi_out_ctls()Takashi Iwai2013-01-121-1/+1
| | | | | | | The multi-io pins are calculated with a blind assumption of cfg->line_outs = 1. This isn't always true. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Simplify the multi-io assignment with multi speakersTakashi Iwai2013-01-121-31/+32
| | | | | | | | | | | | | | | When speakers are chosen as the the primary output during evaluation, we did some tricks to assign the possible multi-io jacks with a certain offset value to multi_out dacs. This was a workaround for the case with multiple speakers like Acer Aspire. But this is quite ugly at the same time and the resultant code is hard to understand. More badly, it works wrongly for 2.1 speakers like Apple iMac91. In this patch, instead of fiddling with the offset to multi_out dacs, simply add a certain badness number if headphone(s) + multi-ios are possible. This simplify the code a bit, and it's more robust. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Check the existing path in snd_hda_add_new_path()Takashi Iwai2013-01-121-7/+24
| | | | | | | If the requested path has been already added, return the existing path instance instead of adding a duplicated instance. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Avoid duplicated path creationsTakashi Iwai2013-01-121-2/+7
| | | | | | | | When the paths are created in map_singles(), we don't have to re-create new paths in try_assign_dacs(). Just evaluate the badness and skip to the next item. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Initialize output paths with current active statesTakashi Iwai2013-01-121-5/+7
| | | | | | | | | | Set path->active flag at the path creation time and let the paths initialized according to the current path->active state in set_output_and_unmute(). This allows to modify the active flag of some output paths dynamically, e.g. switching the front output route with or without aamix like patch_via.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Don't skip amp init for activated pathsTakashi Iwai2013-01-121-1/+1
| | | | | | | | | | | activate_amp() in the generic parser checks whether the given NID is included in any active paths and skips it if found. This was a workaround for avoiding disabling the widgets in the active paths when one path is disabled, thus it shouldn't be applied to the case for path activation. Due to this wrong check, some analog loopback paths haven't been initialized correctly. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add hooks for HP/line/mic auto switchingTakashi Iwai2013-01-121-0/+6
| | | | | | ... as a preliminary work for migrating patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Revive snd_hda_get_conn_list()Takashi Iwai2013-01-121-4/+4
| | | | | | | | | | | | | Manage the connection list cache using linked lists instead of snd_array, and revive snd_hda_get_conn_list() again, so that we don't have to keep the expanded values locally. This will reduce the stack usage by recursive call of snd_hda_get_conn_index() or parse_nid_path() of the generic parser. The list management doesn't include any mutex protection, thus the caller needs to take care of race appropriately. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add inv_eapd flag to struct hda_codecTakashi Iwai2013-01-121-0/+2
| | | | | | | | | Add the new flag, codec->inv_eapd, indicating that the EAPD implementation is inverted. There are always broken hardware in the world. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Implement independent HP controlTakashi Iwai2013-01-121-1/+129
| | | | | | | | | | | | Similar like the implementation in patch_analog.c and patch_via.c, the generic parser can provide the independent HP PCM stream now. It's enabled when spec->indep_hp is set by the caller while parsing. Currently no dynamic PCM switching as in patch_via.c is implemented yet. The control returns -EBUSY when the value is changed during PCM operations. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Allow aamix in the primary output pathTakashi Iwai2013-01-121-0/+7
| | | | | | | Allow the path including the loopback mixer widget in the primary output channel as an alternative in the generic codec parser. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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