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* ALSA: Revert "ALSA: hda/realtek - Avoid invalid COEFs for ALC271X"Martin Kepplinger2014-02-121-3/+0
| | | | | | | | | | | | | | | This reverts commit d3c56568f43807135f2c2a09582a69f809f0d8b7. 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: d3c56568f438 ('ALSA: hda/realtek - Avoid invalid COEFs for ALC271X') Cc: <stable@vger.kernel.org> Tested-by: Martin Kepplinger <martink@posteo.de> Signed-off-by: Martin Kepplinger <martink@posteo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix undefined symbol due to builtin/module mixupTakashi Iwai2014-02-101-15/+15
| | | | | | | | | | | | | | | | | | 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: f8f1becfa4ac ('ALSA: hda - Fix leftover ifdef checks after modularization') Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix mic capture on Sony VAIO Pro 11Takashi Iwai2014-02-071-0/+1
| | | | | | | | | 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: Hendrik-Jan Heins <hjheins@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add a headset quirk for Dell XPS 13David Henningsson2014-02-071-0/+1
| | | | | | | | | | | 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: Po-Hsu Lin <po-hsu.lin@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix inconsistent Mic mute LEDTakashi Iwai2014-02-076-18/+31
| | | | | | | | | | | | | | 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 - Fix leftover ifdef checks after modularizationTakashi Iwai2014-02-072-4/+4
| | | | | | | | | | | | | Since the commit [595fe1b702c3: 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: 595fe1b702c3 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161 Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Improve loopback path lookups for AD1983Takashi Iwai2014-02-051-0/+7
| | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix missing VREF setup for Mac Pro 1,1Takashi Iwai2014-02-051-1/+8
| | | | | | | | | | 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: Nicolai Beuermann <mail@nico-beuermann.de> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Add missing mixer widget for AD1983Takashi Iwai2014-02-051-0/+1
| | | | | | | | | The mixer widget on AD1983 at NID 0x0e was missing in the commit [f2f8be43c5c9: 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: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/realtek - Avoid invalid COEFs for ALC271XTakashi Iwai2014-02-051-0/+3
| | | | | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - Fix silent output on Toshiba Satellite L40Takashi Iwai2014-02-041-0/+19
| | | | | | | | | | | | 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: Takashi Iwai <tiwai@suse.de>
* ALSA: hda/hdmi - allow PIN_OUT to be dynamically enabledStephen Warren2014-01-311-4/+36
| | | | | | | | | | | | | | | | | | | | | | | | Commit 384a48d71520 "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 6169b673618b "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: Uosis <uosisl@gmail.com> Cc: <stable@vger.kernel.org> # v3.13 Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: hda - add headset mic detect quirks for another Dell laptopHui Wang2014-01-301-0/+1
| | | | | | | | | | | | | 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: Doro Wu <fan-cheng.wu@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'xonar-dg' of git://git.alsa-project.org/alsa-kprivate into ↵Takashi Iwai2014-01-309-501/+713
|\ | | | | | | | | | | | | | | | | for-next This completes the hardware support for the Asus Xonar DG/DGX cards, and makes them actually usable. This is v4 of Roman's patch set with some small formatting changes.
| * ALSA: oxygen: Xonar DG(X): cleanup and minor changesRoman Volkov2014-01-293-33/+11
| | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): modify high-pass filter controlRoman Volkov2014-01-291-2/+6
| | | | | | | | | | | | | | | | Change the 'put' function of the high-pass filter control to use the new SPI functions. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): modify input select functionsRoman Volkov2014-01-292-25/+30
| | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): modify capture volume functionsRoman Volkov2014-01-292-8/+26
| | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): use headphone volume controlRoman Volkov2014-01-291-32/+83
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): modify playback output selectRoman Volkov2014-01-293-33/+47
| | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): capture from I2S channel 1, not 2Roman Volkov2014-01-291-1/+1
| | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): move the mixer code into another fileRoman Volkov2014-01-294-364/+411
| | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: modify CS4245 register dumping functionRoman Volkov2014-01-291-3/+4
| | | | | | | | | | | | | | Change the function to read the data from the new shadow buffer. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: modify adjust_dg_dac_routing functionRoman Volkov2014-01-292-24/+22
| | | | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): modify DAC/ADC parameters functionRoman Volkov2014-01-291-20/+38
| | | | | | | | | | | | | | | | When selecting the audio sample rate for CS4245, the MCLK divider should also be changed. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): modify initialization functionsRoman Volkov2014-01-292-65/+42
| | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: Xonar DG(X): add new CS4245 SPI functionsRoman Volkov2014-01-293-4/+73
| | | | | | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: additional definitions for the Xonar DG/DGX cardRoman Volkov2014-01-292-14/+23
| | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: change description of the xonar_dg.c fileRoman Volkov2014-01-291-5/+13
| | | | | | | | | | | | | | Add some additional information in comments and my copyright. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: export oxygen_update_dac_routing symbolRoman Volkov2014-01-291-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: add mute mask for the OXYGEN_PLAY_ROUTING registerRoman Volkov2014-01-291-0/+1
| | | | | | | | | | | | | | | | The Xonar DG/DGX driver needs this mask to mute unnecessary channels. Signed-off-by: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: modify the SPI writing functionRoman Volkov2014-01-292-12/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
| * ALSA: oxygen: add the separate SPI waiting functionRoman Volkov2014-01-291-0/+18
| | | | | | | | | | | | | | | | | | 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: Roman Volkov <v1ron@mail.ru> Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
* | ALSA: hda - Add parameter for dumping processing coefficientsDavid Henningsson2014-01-292-2/+33
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Processing coefficients are often a vital part of the codec's configuration, so dumping them can be important. However, because they are undocumented and secret, we do not want to enable this for all codecs by default. Therefore instead add this as a debugging parameter. I have prepared for codecs that want to enable this by default by the extra dump_coef bitfield, but unsure if we want to do that as long as the (unlikely, but still) race remains. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Fix silent output on MacBook Air 1,1Adrien Vergé2014-01-251-1/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Similarly to other Apple products, MBA 1,1 needs a specific quirk. Pin 0x18 must be set to VREF_50 to have sound output. This was no longer done since commit 1a97b7f, resulting in a mute built-in speaker. This patch corrects the regression by creating a fixup for the MBA 1,1. Fixes: 1a97b7f22774 ("ALSA: hda/realtek - Remove the last static quirks for ALC882") Cc: <stable@vger.kernel.org> [v3.4+] Tested-by: Adrien Vergé <adrienverge@gmail.com> Signed-off-by: Adrien Vergé <adrienverge@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Apply +5dB output amp on ASUS Zenbook UX31ATakashi Iwai2014-01-221-1/+13
| | | | | | | | | | | | | | | | | | ASUS Zenbook UX31A has yet another problem -- softer output level than others. According to the measurement, the peak output difference between 31A and 31E is 5dB. As ALC269VB has a COEF for the class-D pre-amp, let's apply it for +5dB. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: cs46xx: Fix memory leak at destructorTakashi Iwai2014-01-221-0/+1
| | | | | | | | | | | | | | The release of module object itself was forgotten. Spotted by COVERIY CID 1162828. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - add headset mic detect quirks for some Dell machinesHui Wang2014-01-161-1/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When we plug a 3-ring headset on some Dell machines, the headset mic can't be detected, after apply this patch, the headset mic can work well on all those machines. On the machine with the Subsytem ID 0x10280610, if we use ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be detected and work well, but the sound can't be outputed via headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE can fix this problem. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson@canonical.com> Tested-by: David Chen <david.chen@canonical.com> Tested-by: Cyrus Lien <cyrus.lien@canonical.com> Tested-by: Shawn Wang <shawn.wang@canonical.com> Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Fix endless vmaster hook call in thinkpad_helper.cTakashi Iwai2014-01-141-5/+5
| | | | | | | | | | | | | | | | | | The new vmaster hook, update_tpacpi_mute_led(), calls the original vmaster hook, but I forgot to save the original hook function but keep calling the updated one, which of course results in a stupid endless loop. Fixed now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - automute via amp instead of pinctl on some AIO modelsHui Wang2014-01-141-3/+19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On some AIO (All In One) models with the codec alc668 (Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack, the system will switch the output to headphone and set the speaker to automute as well as change the speaker Pin-ctls from 0x40 to 0x00, this will bring loud noise to the headphone. I tried to disable the corresponding EAPD, but it did not help to eliminate the noise. According to Takashi's suggestion, we use amp operation to replace the pinctl modification for the automute, this really eliminate the noise. BugLink: https://bugs.launchpad.net/bugs/1268468 Cc: David Henningsson <david.henningsson@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Apply codec power_filter to FG nodesTakashi Iwai2014-01-134-29/+27
| | | | | | | | | | | | | | 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 set indep_hp flag for old AD codecsTakashi Iwai2014-01-131-8/+8
| | | | | | | | | | | | | | | | | | | | Some old AD codecs don't like the independent HP handling, either it contains a single DAC (AD1981) or it mandates the mixer routing (AD1986A). This patch removes the indep_hp flag for such codecs. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68081 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: Enable CONFIG_ZONE_DMA for smaller PCI DMA masksTakashi Iwai2014-01-111-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | The PCI devices with DMA masks smaller than 32bit should enable CONFIG_ZONE_DMA. Since the recent change of page allocator, page allocations via dma_alloc_coherent() with the limited DMA mask bits may fail more frequently, ended up with no available buffers, when CONFIG_ZONE_DMA isn't enabled. With CONFIG_ZONE_DMA, the system has much more chance to obtain such pages. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221 Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: ctxfi - Work around emu20k1 glitch to prevent buffered sound data lossSarah Bessmer2014-01-101-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Occasionally, on playback stream ringbuffer wraparound, the EMU20K1 hardware will momentarily return 0 instead of the proper current(loop) address. This patch handles that case, fixing the problem of playback position corruption and subsequent loss of buffered sound data, that occurs with some common buffering layout patterns(e.g. multiple simultaneous output streams with differently-sized or non-power-of-2-sized buffers). An alternate means of fixing the problem would be to read the ca register continuously, until two sequential reads return the same value; however, that would be a more invasive change, has performance implications, and isn't necessary unless there are also issues with the value not being updated atomically in regards to individual bits or something similar(which I have not encountered through light testing). I have no EMU20K2 hardware to confirm if the issue is present there, but even if it's not, this change shouldn't break anything that's not already broken. Signed-off-by: Sarah Bessmer <aotos@fastmail.fm> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: Remove memory reservation code from memalloc helperTakashi Iwai2014-01-093-31/+17
| | | | | | | | | | | | | | | | | | | | | | Nowadays we have CMA for obtaining the contiguous memory pages efficiently. Let's kill the old kludge for reserving the memory pages for large buffers. It was rarely useful (only for preserving pages among module reloading or a little help by an early boot scripting), used only by a couple of drivers, and yet it gives too much ugliness than its benefit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda/hdmi - apply all Haswell fix-ups to Broadwell display codecMengdong Lin2014-01-091-6/+9
| | | | | | | | | | | | | | | | | | Broadwell and Haswell have the same behavior on display audio. So this patch defines is_haswell_plus() to include codecs for both Haswell and its successor Broadwell, and apply all Haswell fix-ups to Broadwell. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - add codec ID for Broadwell display audio codecMengdong Lin2014-01-091-0/+2
| | | | | | | | | | | | | | | | This patch adds codec ID (0x80862808) and module alias for Broadwell display codec. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - add device ID for Broadwell display audio controllerMengdong Lin2014-01-091-0/+3
| | | | | | | | | | | | | | | | This patch adds the device ID for Intel Broadwell display HD-Audio controller, and applies Haswell properties to this device. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: hda - Split Thinkpad ACPI-related codeTakashi Iwai2014-01-083-187/+105
| | | | | | | | | | | | | | | | | | | | | | | | Both patch_realtek.c and patch_conexant.c contain the fairy same code snippet for supporting Thinkpad ACPI LED controls. Split them into thinkpad_helper.c and include it from both places. Although this isn't the best approach from the code size POV, the probability for coexistence of both Realtek and Conexant codecs on a single machine is pretty low, thus it'll end up with less memory footprint than splitting to yet another module. 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>
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