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2011-11-30ASoC: Tegra: Tegra3 requires SND_SOC_TEGRA30_AHUBSumit Bhattacharya
Bug 862023 Change-Id: Ie16d410c7021640898bfbc8fcf451d14e832af6f Reviewed-on: http://git-master/r/47155 Tested-by: Sumit Bhattacharya <sumitb@nvidia.com> Reviewed-by: Stephen Warren <swarren@nvidia.com> Reviewed-by: Scott Peterson <speterson@nvidia.com> Reviewed-by: Scott Williams <scwilliams@nvidia.com> Rebase-Id: Raa466f8f365399c0091a173752dfc9d1c83316f8
2011-11-30ASoC: Tegra: Support Cardhu board in tegra_wm8903.cStephen Warren
Cardhu is a Tegra30 tablet reference design. In terms of audio, it is extremely similar to Harmony/Ventana, except that the WM8903 codec is attached to a different I2C bus ID, and different I2S port. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: Re7572cfde2e359dfb8cf16a5eff38e271f25479d
2011-11-30ASoC: Tegra: Add Kconfig and Makefile support for Tegra30Stephen Warren
Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R6c7ea5a4557593b11ab3cbe2da76ba74b19c8efc
2011-11-30ASoC: Tegra: Add tegra30-ahub driverStephen Warren
The AHUB (Audio Hub) is a mux/crossbar which links all audio-related devices except the HDA controller on Tegra30. The devices include the DMA FIFOs, DAM (Digital Audio Mixers), I2S controllers, and SPDIF controllers. Audio data may be routed between these devices in various combinations as required by board design/application. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R5c2d6b0512231268c773be879cabb11a44d181b8
2011-11-30ASoC: Tegra: Add tegra30-i2s driverStephen Warren
This provides an ASoC DAI interface for Tegra 30's I2S controller. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R7ff210416560aa776e2e751ceba765083f4ce6e5
2011-11-30ASoC: Tegra: utils: Add support for Tegra30 devicesStephen Warren
Tegra30 has some additional clocks that need to be manipulated, names some clocks differently, runs PLLs at different base rates, etc. The utility code needs to handle this. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R295313c25d9eae15a7cc423953f50f9132790c9e
2011-11-30ASoC: Tegra: Complete Tegra->Tegra20 renamingStephen Warren
Rename Tegra20-specific Kconfig variables, module filenames, all internal symbol names, clocks, and platform devices, to reflect the fact the DAS and I2S drivers are for a specific HW version. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: Re5b72edd9dff4c93466b2db96dd40eb708343ccb
2011-11-30ASoC: Tegra: Rename Tegra20 I2S/DAS driver filesStephen Warren
Rename these files so they include a specific hardware version in their filenames. The contents is only touched minimally so that git's rename tracking operates correctly; renaming all symbols in the files results in a diff so large that the rename detection fails. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: Reba100e9e3080ff3b1ff455db4dc3ae166468744
2011-11-30ASoC: Tegra: Make Tegra20 drivers depend on Tegra20 supportStephen Warren
Without this, the Tegra20 drivers can be built into a kernel that's built for Tegra30. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R35d2a776e6a245271c77e4c98e9e80f698e4a9aa
2011-11-30ASoC: Tegra: Introduce separate Kconfig variable for DAS driverStephen Warren
This is mainly for symmetry with a future Tegra30 driver, where the equivalent of the DAS (the AHUB) is useful separately from the I2S driver. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R88e3410c1d13524fac081da2f5e39dc0a2161dee
2011-11-30ASoC: Tegra: Remove unnecessary includesStephen Warren
These includes aren't needed, and the files are about to be renamed. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: Ra771bcec753a844849744da8d7c67f403e7a1dcd
2011-11-30CHROMIUM: ASoC: Tegra: Double-buffer DMA in HWStephen Warren
Avoid audio FIFO underflow due to DMA ISR latency. MODE_ONESHOT works like this: * CPU tells DMA HW to make a transfer . * DMA HW performs the transfer, fires an interrupt, and goes idle. * Interrupt latency (including e.g. LP2 exit time) is seen here. * DMA driver ISR programs the next transfer, from a SW transfer queue. In that model, the interrupt latency must be hidden by the I2S FIFO, which is 16 samples IIRC, i.e. a very short time (~0.3mS @ 48KHz). MODE_CONTINUOUS_SINGLE works like this: * CPU tells DMA HW to make a transfer . * CPU tells DMA HW to make a second transfer. * DMA HW performs the transfer, fires an interrupt, and starts the second transfer immediately. * DMA driver ISR programs the next (third) transfer, while the second is in progress, from a SW transfer queue. Thus any DMA ISR latency is seen between transfer N completing and transfer N+2 starting (rather than N+1), since the HW itself already knows how to perform transfer N+1 before transfer N completes. Hence, the system is tolerant of much larger latencies. BUG=chromium-os-partner:3294 TEST=Seaboard clamshell: "aplay sin-500hz-48000-60s.wav" multiple times while system idle and power LED flickering indicating CPU power saving. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Olof Johansson <olofj@chromium.org> Rebase-Id: R4063e2abdf57f47ebe4dc63488056268747bea82
2011-11-30ASoC: Tegra: I2S: Modify clock management logicStephen Warren
Modify the clock management logic to match that implemented in the upcoming Tegra30 ASoC driver. Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R2e5a6c0989f3f91c95cf4e6ad890fc236bcd8fa5
2011-11-30ASoC: Tegra: Conflict with legacy audio APIsStephen Warren
Signed-off-by: Stephen Warren <swarren@nvidia.com> Rebase-Id: R2f3e8ab21f02ee24f59600862729a494ca054bda
2011-11-30arm: tegra: dma: Adding client name with dma allocation.Laxman Dewangan
By changing the dma allocation API to take the client name, it is easy to track who is allocated the DMA channels when we run out of the DMA channels. Original-Change-Id: I016011cfd74089fed0da1bc0f121800017ce124a Reviewed-on: http://git-master/r/28031 Reviewed-by: Varun Colbert <vcolbert@nvidia.com> Tested-by: Varun Colbert <vcolbert@nvidia.com> Original-Change-Id: I048bcb87f95ee6d8ad2fdce993a1758dc5071666 Rebase-Id: Rf6d68d90cb4a8e91be3ec921cadc116f3a36184c
2011-11-26ALSA: hda - fix ELD memory leakWu Fengguang
Backported from commit b95d68b8179764e29558b75cec35ef4a6a98925b. memset(eld) clears eld->proc_entry which will leak the struct snd_info_entry when unloading module. Fix it by - memset only the fields before eld->eld_buffer - set eld->eld_valid to true _after_ all eld fields have been filled Cc: Pierre-louis Bossart <pierre-louis.bossart@intel.com> Acked-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-26ALSA: hda - Fix the connection selection of ADCs on Cirrus codecsTakashi Iwai
commit 05ee7964a470d29889ac48cc8274c1b5a1904a11 upstream. spec->cur_adc isn't set until cs_capture_pcm_prepare() is called although the driver tries to select the connection at init time and at auto-mic switch. This results in the access to the widget NID 0, which is obviously invalid, also a wrong capture source. This patch fixes the issue by issuing the connect-select verb conditionally at appropriate places. Reported-and-tested-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: usb-audio - Fix the missing volume quirks at delayed initTakashi Iwai
commit dcaaf9f2c16b56f8bb316881fcd3f15c18fc71e7 upstream. In the recent usb-audio driver, the initialization of volume ranges may be delayed when the device doesn't respond well at the probing time. But the volume quirks for certain devices are applied only in mixer_ctl_feature_info() thus only at the very first probe and will be missing when the volume range is initialized later. This patch moves the volume quirk code to be always called from the volume-range extraction (get_min_max()), so that the quirks are properly applied in the later init time. Reported-and-tested-by: Alexey Fisher <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: usb-audio - Check the dB-range validity in the later read, tooTakashi Iwai
commit 9fcd0ab130579d9742538340edda3225f2b49a3e upstream. When the initial check of dB-range failed due to the read error, try to check again at the later read, too. When an invalid dB range is found, remove TLV flags and notify the mixer info change. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ASoC: Don't use wm8994->control_data in wm8994_readable_register()Mark Brown
commit 8eeea521d9d0fa6afd62df8c6e6566ee946117fa upstream. The field is no longer initialised so this will crash if running on wm8958. Reported-by: Thomas Abraham <thomas.abraham@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: hda - Don't add elements of other codecs to vmaster slaveTakashi Iwai
commit aeb4b88ec0a948efce8e3a23a8f964d3560a7308 upstream. When a virtual mater control is created, the driver looks for slave elements from the assigned card instance. But this may include the elements of other codecs when multiple codecs are on the same HD-audio bus. This works at the first time, but it'll give Oops when it's once freed and re-created via reconfig sysfs. This patch changes the element-look-up strategy to limit only to the mixer elements of the same codec. Reported-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: hda - fix internal mic on Dell Vostro 3500 laptopJulian Wollrath
commit f7f9bdfadfda07afb904a9767468e38c2d1a6033 upstream. Fix the not working internal mic on Dell Vostro 3500 laptop by introducing the new model dell-vostro-3500. Signed-off-by: Julian Wollrath <jwollrath@web.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Ensure we always delay for WM8962 FLL when starting from SYSCLKMark Brown
commit 0b7dd6ad92b6cace35dc5d06d6e236c2751c85dc upstream. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Ensure the WM8962 oscillator and PLLs start up disabledMark Brown
commit 2af8de8c39cf58e5a5e40a9d5d71332da98e6ba7 upstream. Since there is no current software control for these they would otherwise be left enabled, consuming power. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Ensure WM8962 PLL registers are resetMark Brown
commit 4f4488abc97c1c27ff029f887944e6a6da1f5733 upstream. The WM8962 has a separate software reset for the PLL registers. Ensure that these are reset also on startup. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: WM8904: Set `invert' bit for Capture SwitchHong Xu
commit 5a7c5f26df3c0122814dfa1c13ef6dfbdbffdb86 upstream. Set `invert' bit for Capture Switch. Otherwise analogue is muted when Capture Switch is ON. Signed-off-by: Hong Xu <hong.xu@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Leave input audio data bit length settings untouched in wm8711_set_dai_fmtAxel Lin
commit d558cfc30064a97c2c65dbd2b3a4f5a1dea7ec1b upstream. Current implementation in wm8711_set_dai_fmt always clear BIT[3:2] (the Input Audio Data Bit Length Select) of WM8711_IFACE(07h) register. Input Audio Data Bit Length Select bits are set by wm8711_hw_params, we should leave BIT[3:2] untouched in wm8711_set_dai_fmt. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8711: Fix wrong mask for setting input audio data bit length selectAxel Lin
commit 04c57163c8edfbc50e022737014069998ba4fc5f upstream. The Input Audio Data Bit Length Select is controlled by BIT[3:2] of WM8711_IFACE(07h) register. Current code incorrectly masks BIT[1:0] which is for Audio Data Format Select. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Fix a regression for DMA-position check with CA0110Takashi Iwai
commit 69f9ba9b0cad67bc03f0a096f7f274de795ca844 upstream. The regression-fix in 3.1 for the check of DMA-position validity caused yet another regression for CA0110. As usual, this hardware seems working only with LPIB properly. Adding the appropriate driver-caps bit to force LPIB fixes the problem. Reported-and-tested-by: Andres Freund <andres@anarazel.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Fix silent output regression with ALC861Takashi Iwai
commit 43dea228a3ba5463392281535dfb3d3fe56f4c2c upstream. The 3.1 kernel has a regression for ALC861 codec where no sound output is heard with the default setup. It's because the amps in DACs aren't properly unmuted while the output mixers are assigned only to pins. This patch fixes the missing initialization of DACs when no mixer is assigned to them. Tested-by: Andrea Iob <andrea_iob@yahoo.it> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda/realtek - Skip invalid digital out pinsTakashi Iwai
commit 51e4152a969aa6d2306492ebf143932dcb535c9b upstream. Some BIOS report invalid pins as digital output pins. The driver checks the connection but it doesn't do it fully correctly, and it leaves some undefined value as the audio-out widget, which makes the driver spewing warnings. This patch fixes the issue. Reference: https://bugzilla.novell.com/show_bug.cgi?id=727348 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda/realtek - Don't create alt-stream for capture when unnecessaryTakashi Iwai
commit 1fa1757366783fb52e6e85c2d735db49b818d382 upstream. When the driver finds multiple ADCs, it tries to create an alternative capture PCM stream. However, these secondary ADCs might be useless or in uncontrolled paths in some cases, e.g. when auto-mic or dynamic ADC-switching is enabled. Also, when only a single capture source is available, the multi-streams don't make sense, too. With this patch, the driver checks such condition and skips the alt stream appropriately. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Add support for 92HD65 / 92HD66 family of codecsCharles Chin
commit ad5d8755116b431f0709c745ee17cb567a478d43 upstream. These codecs have SPDIF-in, which is new to the 92HD83xxx compatible families, so a bit of logic is added to support them. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Disable power-widget control for IDT 92HD83/93 as defaultCharles Chin
commit 35c11777b906042eca9e6f1c03e464726c7faa07 upstream. The power-widget control in patch_stac92hd83xxx() never worked properly, thus it's safer to turn it off as default for now. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: ua101: fix crash when unpluggingClemens Ladisch
commit 862a6244eb9f9f5123fe819454fcfcae0ee1f2f9 upstream. If the device is unplugged while running, it is possible for a PCM device to be closed after the disconnect callback has returned. This means that kill_stream_urb() and disable_iso_interface() would try to access already-invalid or freed USB data structures. The function free_usb_related_resources() was intended to prevent this, but forgot to clear the affected variables. Reported-and-tested-by: Olivier Courtay <olivier@courtay.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Add missing static ADC tables for ALC269 quirksTakashi Iwai
[There is no upstream commit for this patch since the corresponding code was removed from 3.2 kernel. This is a regression found only in 3.1 kernel, so please apply this only to 3.1.x series.] Some ALC269 quirks define their own .cap_mixer field but without the static adc_nids[]. This resulted in the mismatch of ADC because ALC269 may have the widget 0x07 for another audio-in, and the auto-parser picks this up instead. This patch fixes the problem by adding the static adc_nids[] and co again to these entries. Tested-by: Chris Vine <chris@cvine.freeserve.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Fix ADC input-amp handling for Cx20549 codecTakashi Iwai
commit 6b45214277bec2193ad3ccb8d7aa6100b5a0f1a9 upstream. It seems that Conexant CX20549 chip handle only a single input-amp even though the audio-input widget has multiple sources. This has been never clear, and I implemented in the current way based on the debug information I got at the early time -- the device reacts individual input-amp values for different sources. This is true for another Conexant codec, but it's not applied to CX20549 actually. This patch changes the auto-parser code to handle a single input-amp per audio-in widget for CX20549. After applying this, you'll see only a single "Capture" volume control instead of separate "Mic" or "Line" captures when the device is set up to use a single ADC. We haven't tested 20551 and 20561 codecs yet. If these show the similar behavior like 20549, they need to set spec->single_adc_amp=1, too. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Fix a bug in WM8962 DSP_A and DSP_B settingsSusan Gao
commit fbc7c62a3ff831aef24894b7982cd1adb2b7e070 upstream. Signed-off-by: Susan Gao <sgao@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmico.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Remove direct register cache accesses from WM8962 driverMark Brown
commit 38f3f31a0a797bdbcc0cdb12553bbecc2f9a91c4 upstream. Also fix return values for speaker switch updates. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 CaptureAxel Lin
commit 35024f4922f7b271e7529673413889aa3d51c5fc upstream. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8994: Fix setting rate_reg for wm8994-aif2Axel Lin
commit c527e6aadc8f142ad388b6aa59a1ce6a4bfb1966 upstream. For wm8994-aif2, the rate_reg should be WM8994_AIF2_RATE. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2Axel Lin
commit 3205e6629bc0eb747fb7d1b4b8fec00b7b919e58 upstream. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: ak4535: fixup cache register tableAxel Lin
commit 7c04241acbdaf97f1448dcccd27ea0fcd1a57684 upstream. ak4535_reg should be 8bit, but cache table is defined as 16bit. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: ak4642: fixup cache register tableKuninori Morimoto
commit 19b115e523208a926813751aac8934cf3fc6085e upstream. ak4642 register was 8bit, but cache table was defined as 16bit. ak4642 doesn't work correctry without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8741: Fix setting interface format for DSP modesAxel Lin
commit 3a340104fad6ecbea5ad6792a2ea855f0507a6e0 upstream. According to the datasheet: Format Control (05h) BITS[3:2] FMT[1:0] Audio data format selection 00 = right justified mode 01 = left justified mode 10 = I2S mode 11 = DSP mode BIT[4] LRP Polarity selec for LRCLK/DSP mode select 0 = normal LRCLK poalrity/DSP mode A 1 = inverted LRCLK poarity/DSP mode B For SND_SOC_DAIFMT_DSP_A, we should set 0x000C instead of 0x0003. For SND_SOC_DAIFMT_DSP_B, we should set 0x001C instead of 0x0013. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8940: Properly set codec->dapm.bias_levelAxel Lin
commit 5927f94700e860ae27ff24e7f3bc9e4f7b9922eb upstream. Reported-by: Chris Paulson-Ellis <chris@edesix.com> Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: HDA: Add new revision for ALC662David Henningsson
commit cc667a72d471e79fd8e5e291ea115923cf44dca0 upstream. The revision 0x100300 was found for ALC662. It seems to work well with patch_alc662. BugLink: http://bugs.launchpad.net/bugs/877373 Tested-by: Shengyao Xue <Shengyao.xue@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Acked-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Keep EAPD turned on for old Conexant chipsTakashi Iwai
commit 254f296840b64b034a4c850d45dbde7c040f0819 upstream. In the old Conexant chips (5045, 5047, 5051 and 5066), a single EAPD may handle both headphone and speaker outputs while it's assigned only to one of them. Turning off dynamically leads to the unexpected silent output in such a configuration with the auto-mute function. Since it's difficult to know how the EAPD is handled in the actual h/w implementation, better to keep EAPD on while running for such codecs. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Remove bad code for IDT 92HD83 family patchCharles Chin
commit 6c5c04e509b7000617b09d4301f0b9b6d171d1e6 upstream. The purpose of this patch is to remove a section of "bad" code that assigns the last DAC to ports E or F in order to support notebooks with docking in earlier days, around ALSA 1.0.19 - 21. This is not necessary now and actually breaks some configurations that use these ports as other devices. This have been tested on several different configurations to make sure that it is working for different combinations. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-10-21Merge branch 'fix/hda' of git://github.com/tiwai/soundLinus Torvalds
* 'fix/hda' of git://github.com/tiwai/sound: ALSA: HDA: conexant support for Lenovo T520/W520 ALSA: hda - Add position_fix quirk for Dell Inspiron 1010
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