- 04 8月, 2011 4 次提交
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由 Pierre-Louis Bossart 提交于
Add new parameter to disable rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and prevents users from specifying exact period/buffer sizes. For example for 44.1kHz, a period size set to 20ms will be rounded to 19.59ms. Tested and enabled on Intel HDA controllers. Option is disabled by default for other controllers. Tested-by: NWu Fengguang <fengguang.wu@intel.com> Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Add a few tracepoints to HD-audio driver. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
It works fine with auto-parser and now the digital mic workaround was implemented in auto-parser fixup, let's drop the static model quirks for these models. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The digital-mic unit on ASUS Eee PC gives PDM signals instead of the normal stereo PCM, thus you can't record a mono stream from the stereo stream as is; the summed stereo signal results in almost zero level, and you'll hear only soft noise. As a workaround, use ALC269-specific COEF to manipulate the dmic route for mono, like used for ALC271x. This is implemented as a fix-up, thus it works only with model=auto or without REALTEK_QUIRKS Kconfig. Reported-and-tested-by: NPavel Roskin <proski@gnu.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 27 7月, 2011 3 次提交
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由 Takashi Iwai 提交于
Copying hp_pins and speaker_pins from line_out_pins may confuse the parser, and it can lead to duplicated initializations for the same pin with a wrong DAC assignment. The problem appears in 3.0 kernel code. Cc: <stable@kernel.org> (for 3.0) Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Somce quirk models don't set adc_nids but let the parser filling it. But the recent code has unnecessary NULL-checks of spec->input_mux, and it resulted in NULL dereferences. This patch fixes that regression. Reported-and-tested-by: NOliver Neukum <oneukum@suse.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Vitaliy Kulikov 提交于
This patch also registers all necessary callbacks to support mute LED only when such control is enabled. And it keeps codec AFG in D0 or D1 state all the time when aggressive power managemnt is enabled for vref-out control (and mute LED) work correctly. Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 7月, 2011 6 次提交
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由 Tim Howe 提交于
This update includes the changes necessary for supporting the CS421x family of codecs. Previously this file only supported the CS420x family of codecs. This file also contains init verbs to correct several issues in the CS421x hardware. Behavior between the CS421x and CS420x codec families is similar, so several functions have been reused with "if" statements to determine which codec family (CS421x or CS420x) is present. Also, this file will be updated sometime in the near future in order to add support for a system using CS421x that requires mono mix on the speaker output only. [Fix const usages and adaption for new APIs by tiwai] Signed-off-by: NTim Howe <tim.howe@cirrus.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The procedure for codec D-state change may have exceptional cases depending on the codec chip, such as a longer delay or suppressing D3. This patch adds a new codec ops, set_power_state() to override the system default function. For ease of porting, snd_hda_codec_set_power_to_all() helper function is extracted from the default set_power_state() function. As an example, the Conexant codec-specific delay is removed from the default routine but moved to patch_conexant.c. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Add a new ops, post_suspend(), which is called after suspend() ops is performed. This is called only in the case of the real PM suspend, and the codec driver can use this for further changing of D-state or clearing the LED, etc. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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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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由 Vitaliy Kulikov 提交于
This patch adds checking of mute state on all outputs besides just speakers to calculate the master mute state for mute led support. It also renames and splits the function that does it for better code clarity. Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Vitaliy Kulikov 提交于
Codec state is not restored immediately on resume but on the first access when power-save is enabled. That leads to an invalid mute led state after resume until either sound is played or some control is changed. This patch adds a possibility for a vendor specific patch to restore codec state immediately after resume if required. And it adds code to restore IDT codecs state immediately on resume on HP systems with mute led support. Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 24 7月, 2011 2 次提交
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由 Vitaliy Kulikov 提交于
Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Fix a regression in the DAC filling code in patch_realtek.c. The already filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0, thus always pointed to the first DAC. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 7月, 2011 2 次提交
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由 Takashi Iwai 提交于
"force" argument is always true, so let's strip it off. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
This patch fixes non-working indep-HP control on VT1708* codecs. The problems are that via_independent_hp_put() wasn't fixed to follow the recent change of three HP paths, and hp_indep_path didn't contain the amp nids of mixer elements. Together with the fixes, a few code clean-ups are done. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 7月, 2011 2 次提交
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由 Takashi Iwai 提交于
This patch changes the behavior of independent-HP enum switch. Now instead of returning a busy error, the driver switches dynamically the stream of the HP (and shared) DACs according to the current mode. The logic is similar like the dual-mic ADC switch, but a bit more complicated because of the presence of shared DAC. Together with the change, a mutex is introduced to protect against the possible races for the indep-HP mode setting. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
This patch adds the dynamic control of analog-loopback for VIA codecs. When the loopback is enabled, the inputs from line-ins and mics are mixed with the front DAC, and sent to the front outputs. The very same input is routed to the headhpones and speakers in loopback mode. However, since the loopback mix can't take other than the front DAC, there is no longer individual volume controls for headphones and speakers. Once when the loopback control is off, these volumes take effect. Since the individual volumes are more desired in general use caess, the loopback mode is set to off as default for now. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 14 7月, 2011 2 次提交
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由 Takashi Iwai 提交于
It was obviously wrong, grr.... Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
During the rewrite, the check of spec->need_dac_fix and the corresponding num_dacs change was dropped from the channel-mode control. This patch re-adds it, and also enables need_dac_fix for ALC880 as default, as this feature was originally introduced to fix h/w bugs of this chip. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 7月, 2011 1 次提交
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由 Paul Menzel 提交于
This commit is a fix up for commit acfa634f. commit acfa634f Author: Takashi Iwai <tiwai@suse.de> Date: Tue Jul 12 17:27:46 2011 +0200 ALSA: hda - Add Kconfig for the default buffer size Signed-off-by: NPaul Menzel <paulepanter@users.sourceforge.net> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 12 7月, 2011 5 次提交
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由 Takashi Iwai 提交于
Add a Kconfig entry to specify the default buffer size. Distros using PulseAudio can choose a larger value here. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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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
VT1718S and co have a secret connection from DAC to AA-mix, which doesn't appear in the connection list obtained from the h/w. Currently the driver fixes the connection index locally at init, but now we can expose it statically via snd_hda_override_connections() so that this conection can be checked better by the parser in future. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
In the codec proc outputs, read the raw connections instead of the cached connection list, i.e. proc files contain only raw values. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Add a function to add/modify the connection-list cache entry. It'll be useful to fix a buggy hardware result. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 11 7月, 2011 4 次提交
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由 Takashi Iwai 提交于
Some machines seem to use EAPD control of the unused pin for controlling the overall EAPD. Since the driver currently doesn't check the EAPD of unused pins, the EAPD isn't enabled. For avoiding such a problem, turn all extra EAPDs on as default. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
For smart51 pins, we need to preserve the input pin-control bits at auto-mute controls instead of overwriting zero or pin-out-only. Otherwise the VREF won't be set properly when smart51 is disabled again. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
When Independent-HP mode is changed for VIA, the driver needs to re-issue the auto-mute check so that the line-out pins are set properly without influence of HP pin state. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
When the line-jack is plugged/unplugged, the driver must check also the headphone jack state in addition to the line-out jack. Currently it checks only the line-out state and ignores the headphone. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 09 7月, 2011 2 次提交
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由 Takashi Iwai 提交于
Instead of checking the model quirk, use a fixup table for workaround of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
It's harmless but annyoing. sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’: sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 08 7月, 2011 7 次提交
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由 Takashi Iwai 提交于
Now all alc*_parse_auto_config() do almost same thing except for the NID list to ignore and the PINs for SSID-check, we can merge all these to a single function. A good amount of code reduction. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Finally the last one. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
One more code reduction. This codec has less DACs, thus the wiring to DAC can't be filled uniquely for all output pins, i.e. some outputs share the same volume control. Except for that, all seems working fine. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Now coming to ALC268/269 parser codes. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Merge more auto-parser code in patch_realtek.c, now for ALC861. The topology of this codec is pretty simple, and can be parsed well by the current starndard parser. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
query_amp_caps() may return non-zero if the amp cap isn't supported by the codec. Thus one needs to check widget-caps first, then check the corresponding amp-caps. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The existing standard auto-parser can work well with this codec, too. Let's merge. Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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