- 03 3月, 2014 1 次提交
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由 Marius Knaust 提交于
Signed-off-by: NMarius Knaust <marius.knaust@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 26 2月, 2014 1 次提交
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由 Kailang Yang 提交于
I lost this SSID. Add it into the fixup table. Signed-off-by: NKailang Yang <kailang@realtek.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 25 2月, 2014 1 次提交
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由 Takashi Iwai 提交于
HP Folio 13 may have a broken BIOS that doesn't set up the mute LED GPIO properly, and the driver guesses it wrongly, too. Add a new fixup entry for setting the GPIO pin statically for this laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70991 Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 2月, 2014 1 次提交
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由 Kailang Yang 提交于
More HP machine need mute led support. Signed-off-by: NKailang Yang <kailang@realtek.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 20 2月, 2014 1 次提交
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由 Hui Wang 提交于
The front headphone and mic jackes on a HP desktop model (Vendor Id: 0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this machine has 8 physical ports, 6 of them are routed to rear jackes and all of them work very well, while the remaining 2 ports are routed to front headphone and mic jackes, but the corresponding pin complex node are not defined correctly. After apply this fix, the front audio jackes can work very well. [trivial fix of enum definition by tiwai] BugLink: https://bugs.launchpad.net/bugs/1282369 Cc: David Henningsson <david.henningsson@canonical.com> Tested-by: NGerald Yang <gerald.yang@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: NHui Wang <hui.wang@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 19 2月, 2014 2 次提交
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由 Hsin-Yu Chao 提交于
Incorrect ADC is picked in ca0132_capture_pcm_prepare(), where it assumes multiple streams while there is one stream per ADC. Note that ca0132_capture_pcm_cleanup() already does the right thing. The Chromebook Pixel has a microphone under the keyboard that is attached to node id 0x8. Before this fix, recording would always go to the main internal mic (node id 0x7). Signed-off-by: NHsin-Yu Chao <hychao@chromium.org> Reviewed-by: NDylan Reid <dgreid@chromium.org> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Hsin-Yu Chao 提交于
When a HDMI stream is opened with the same stream tag as a following opened stream to ca0132, audio will be heard from two ports simultaneously. Fix this issue by change to use snd_hda_codec_setup_stream and snd_hda_codec_cleanup_stream instead, so that an inactive stream can be marked as 'dirty' when found with a conflict stream tag, and then get purified. Signed-off-by: NHsin-Yu Chao <hychao@chromium.org> Reviewed-by: NChih-Chung Chang <chihchung@chromium.org> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 18 2月, 2014 1 次提交
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由 Hui Wang 提交于
When we plug a 3-ring headset on the Dell machines (Vendor ID: 0x10ec0255, Subsystem ID: 0x10280657; Vendor ID: 0x10ec0255, Subsystem ID: 0x1028065f), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson@canonical.com> Tested-by: NCyrus Lien <cyrus.lien@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: NHui Wang <hui.wang@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 13 2月, 2014 1 次提交
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由 Martin Kepplinger 提交于
This reverts commit d3c56568. The reverted commit breaks audio through headphone line out on the Acer TravelMate B113 (Type1Sku0) Notebook, my main work machine. I don't know much about it but this fixes my problem. Bisected and tested. Fixes: d3c56568 ('ALSA: hda/realtek - Avoid invalid COEFs for ALC271X') Cc: <stable@vger.kernel.org> Tested-by: NMartin Kepplinger <martink@posteo.de> Signed-off-by: NMartin Kepplinger <martink@posteo.de> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 10 2月, 2014 1 次提交
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由 Takashi Iwai 提交于
Even after the fix for leftover kconfig handling (commit f8f1becf), the current code still doesn't handle properly the builtin/module mixup case between the core snd-hda-codec and other codec drivers. For example, when CONFIG_SND_HDA_INTEL=y and CONFIG_SND_HDA_CODEC_HDMI=m, it'll end up with an unresolved symbol snd_hda_parse_hdmi_codec. This patch fixes the issue. Now codec->parser points to the parser object *only* when a module (either generic or HDMI parser) is loaded and bound. When a builtin symbol is used, codec->parser still points to NULL. This is the difference from the previous versions. Fixes: f8f1becf ('ALSA: hda - Fix leftover ifdef checks after modularization') Reported-by: NFengguang Wu <fengguang.wu@intel.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 07 2月, 2014 4 次提交
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由 Takashi Iwai 提交于
The very same fixup is needed to make the mic on Sony VAIO Pro 11 working as well as VAIO Pro 13 model. Reported-and-tested-by: NHendrik-Jan Heins <hjheins@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 David Henningsson 提交于
This quirk is needed for the headset microphone to work. Alsa-info at http://www.alsa-project.org/db/?f=8c7dfe857ceff462ca2de133e67023c0f68de9cb Cc: stable@vger.kernel.org (3.10+) Reported-by: NPo-Hsu Lin <po-hsu.lin@canonical.com> Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
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: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Since the commit [595fe1b7: ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate], the kconfig variables for the generic parser and codec drivers can be "m" instead of boolean, but some codes are left unchanged to check only #ifdef CONFIG_SND_HDA_CODEC_XXX, which is no longer true for modules. This patch fixes them by replacing with IS_ENABLED() macros. Fixes: 595fe1b7 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 05 2月, 2014 4 次提交
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由 Takashi Iwai 提交于
AD1983 has flexible loopback routes and the generic parser would take wrong path confusingly instead of taking individual paths via NID 0x0c and 0x0d. For avoiding it, limit the connections at these widgets so that the parser can think more straightforwardly. This fixes the regression of the missing line-in loopback on Dell machine. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011 Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
Mac Pro 1,1 with ALC889A codec needs the VREF setup on NID 0x18 to VREF50, in order to make the speaker working. The same fixup was already needed for MacBook Air 1,1, so we can reuse it. Reported-by: NNicolai Beuermann <mail@nico-beuermann.de> Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
The mixer widget on AD1983 at NID 0x0e was missing in the commit [f2f8be43: ALSA: hda - Add aamix NID to AD codecs]. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011 Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Takashi Iwai 提交于
We've seen often problems after suspend/resume on Acer Aspire One AO725 with ALC271X codec as reported in kernel bugzilla, and it turned out that some COEFs doesn't work and triggers the codec communication stall. Since these magic COEF setups are specific to ALC269VB for some PLL configurations, the machine works even without these manual adjustment. So, let's simply avoid applying them for ALC271X. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Cc: <stable@vger.kernel.org> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 04 2月, 2014 1 次提交
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由 Takashi Iwai 提交于
Toshiba Satellite L40 with AD1986A codec requires the EAPD of NID 0x1b to be constantly on, otherwise the output doesn't work. Unlike most of other AD1986A machines, EAPD is correctly implemented in HD-audio manner (that is, bit set = amp on), so we need to clear the inv_eapd flag in the fixup, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=67481 Cc: <stable@vger.kernel.org> [v3.11+] Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 01 2月, 2014 1 次提交
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由 Stephen Warren 提交于
Commit 384a48d7 "ALSA: hda: HDMI: Support codecs with fewer cvts than pins" dynamically enabled each pin widget's PIN_OUT only when the pin was actively in use. This was required on certain NVIDIA CODECs for correct operation. Specifically, if multiple pin widgets each had their mux input select the same audio converter widget and each pin widget had PIN_OUT enabled, then only one of the pin widgets would actually receive the audio, and often not the one the user wanted! However, this apparently broke some Intel systems, and commit 6169b673 "ALSA: hda - Always turn on pins for HDMI/DP" reverted the dynamic setting of PIN_OUT. This in turn broke the afore-mentioned NVIDIA CODECs. This change supports either dynamic or static handling of PIN_OUT, selected by a flag set up during CODEC initialization. This flag is enabled for all recent NVIDIA GPUs. Reported-by: NUosis <uosisl@gmail.com> Cc: <stable@vger.kernel.org> # v3.13 Signed-off-by: NStephen Warren <swarren@nvidia.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 30 1月, 2014 20 次提交
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由 Hui Wang 提交于
When we plug a 3-ring headset on the Dell machine (Vendor ID: 0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson@canonical.com> Tested-by: NDoro Wu <fan-cheng.wu@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: NHui Wang <hui.wang@canonical.com> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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由 Roman Volkov 提交于
Remove old SPI control functions, change anti-pop init sequence, remove some garbage from structures. The 'Apply' functions must be called at the mixer initialization, otherwise mixer settings sometimes will not be applied at startup. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Change the 'put' function of the high-pass filter control to use the new SPI functions. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
First of all, we should not touch the GPIOs. They are not for selecting the capture source, but they seems just enable the whole audio input curcuit. The 'put' function calls the 'apply' functions to change register values. Change the order of capture sources. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Modify the input_vol_* functions to use the new SPI routines, There is a new applying function that will be called when the capture source changed. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
I tried both variants: volume control and impedance selector. In the first case one minus is that we can't change the volume of multichannel output without additional software volume control. However, I am using this variant for the last three months and this seems good. All multichannel speaker systems have internal amplifier with the volume control included, but not all headphones have this regulator. In the second case, my software volume control does not save the value after reboot. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Change the order of elements in the output select control. This will reduce the number of relay switches. Change 'put' function to call the oxygen_update_dac_routing() function. Otherwise multichannel playback does not work. Also there is a new function to apply settings, this prevents from duplicating the code. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Actually CS4245 connected to the I2S channel 1 for capture, not channel 2. Otherwise capturing and playback does not work for CS4245. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Moving the mixer code away makes things easier. The mixer will control the driver, so the functions of the driver need to be non-static. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Change the function to read the data from the new shadow buffer. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
When selecting the audio output destinations (headphones, FP headphones, multichannel output), the channel routing should be changed depending on what destination selected. Also unnecessary I2S channels are digitally muted. This function called when the user selects the destination in the ALSA mixer. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
When selecting the audio sample rate for CS4245, the MCLK divider should also be changed. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Change CS4245 initialization: different sequence and GPIO values, according to datasheets and reverse-engineering information. Change cleanup/resume/suspend functions, since they use initialization. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Add the new SPI write and read functions. The SPI read function is used for creating initial registers dump and may be used for debugging purposes. SPI operations are cached, so there is a new function to manage the cache (shadow). I have to remove the shift from the CS4245_SPI_* constants, since when we are performing the reading, we need to shift by 8 instead of 16. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Add additional constants to the xonar_dg.h file: capture and playback sources. Move GPIO_* constants and the dg struct to the header file from the xonar_dg.c file. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Add some additional information in comments and my copyright. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
When the user switches the output from stereo to multichannel or vice versa, the driver needs to update the channel routing. Instead of creating additional subroutines, I better export existing oxygen_update_dac_routing symbol from the oxygen mixer and call this function. It calls model.adjust_dac_routing() and my function does the work. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
The Xonar DG/DGX driver needs this mask to mute unnecessary channels. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
Modify the oxygen_write_spi() function to use the newly introduced oxygen_wait_spi() function. Change return value from void to int, so it can return error codes. Older drivers just ignore that return value, new drivers can check this value. We need to wait AFTER initiating the SPI transaction, otherwise read operation will not work. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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由 Roman Volkov 提交于
The oxygen_wait_spi() function now performs waiting when the SPI bus completes a transaction. Introduce the timeout error checking and increase timeout to 200 from 40. Signed-off-by: NRoman Volkov <v1ron@mail.ru> Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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