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2011-01-12ALSA: hda - Add static_hdmi_pcm option to HDMI codec parserTakashi Iwai1-1/+6
The dynamic PCM restriction based on ELD information may lead to the problem in some cases, e.g. when the receiver is turned off. Then it may send a TV HDMI default such as channels = 2. Since it's still plugged, the driver doesn't know whether it's the right configuration for future use. Now, when an app opens the device at this moment, then turn on the receiver, the app still sends channels=2. The right solution is to implement some kind of notification and automatic re-open mechanism. But, this is a goal far ahead. This patch provides a workaround for such a case by providing a new module option static_hdmi_pcm for snd-hda-codec-hdmi module. When this is set to true, the driver doesn't change PCM parameters per ELD information. For users who need the static configuration like the scenario above, set this to true. The parameter can be changed dynamically via sysfs, too. Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
2011-01-12ALSA: hda - Don't refer ELD when unpluggedTakashi Iwai1-1/+1
When unplugged, we shouldn't refer to ELD information for PCM open any more. Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
2011-01-12Merge branch 'for-2.6.38' of ↵Takashi Iwai8-169/+234
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
2011-01-12ASoC: tpa6130a2: Fix compiler warningPeter Ujfalusi1-1/+0
sound/soc/codecs/tpa6130a2.c: In function 'tpa6130a2_add_controls': sound/soc/codecs/tpa6130a2.c:342: warning: unused variable 'dapm' Introduced by commit 39646871a47fd8808c08de0ce7d7ca8393af2805 ("ASoC: tpa6130a2: Replace DAPM code with direct interface"). The DAPM code has been removed from the driver, but the dapm struct remained. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-12ASoC: tlv320dac33: Add DAPM selection for LOM invertPeter Ujfalusi1-5/+50
The L/R LOM line can be invertined side of the corresponding DAC, or inverted from the corresponding LOP. Add control for user space to select the source of the LOM inversion. When only the analog bypass is enabled, and the LOM is inverted from DAC output, we need to power the corresponding DAC. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-12ASoC: DMIC codec: Adding a generic DMIC codecDavid Lambert3-0/+86
This codec is to be used by the DMIC driver to control the DMIC codec. This driver will be used on future implementations of the DMIC driver to support codec specific features. At this time, the codec driver just registers the codec DAI. Signed-off-by: David Lambert <dlambert@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-11ALSA: snd-usb-us122l: Fix missing NULL checksKarsten Wiese1-21/+20
Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11ALSA: snd-usb-us122l: Fix MIDI outputKarsten Wiese1-2/+9
The US-122L always reads 9 bytes per urb unless they are set to 0xFD. Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11ASoC: soc-cache: Fix invalid memory access during snd_soc_lzo_cache_sync()Dimitris Papastamos1-1/+1
The size of the lzo syncing bitmap was incorrectly set to the size of the cache times the word size, however, the correct size is the size of the cache. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-11ASoC: Fix section mismatch in wm8995.cTakashi Iwai1-1/+1
__devinitconst can't be used for data referred in driver struct. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11ALSA: oxygen: add S/PDIF source selection for Claro cardsClemens Ladisch1-4/+88
Add a mixer control to switch between the optical and coaxial S/PDIF inputs on the HT-Omega Claro and Claro halo cards. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11ALSA: oxygen: fix CD/MIDI for X-Meridian (2G)Clemens Ladisch1-0/+3
Enable the X-Meridian's CD input and the X-Meridian 2G's potential MIDI ports. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11ASoC: fix migor audio buildGuennadi Liakhovetski1-1/+1
Commit 6d803ba736abb5e122dede70a4720e4843dd6df4 "ARM: 6483/1: arm & sh: factorised duplicated clkdev.c" broke compilation of migor audio. Use the correct header to fix the problem. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-11ALSA: include delay.h for msleep in Xonar DG supportStephen Rothwell1-0/+1
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ASoC: cs4270: use the built-in register cache supportTimur Tabi1-113/+48
Update the CS4270 driver to use ASoC's internal codec register cache feature. This change allows ASoC to perform the low-level I2C operations necessary to read the register cache. Support is also added for initializing the register cache with an array of known power-on default values. The CS4270 driver was handling the register cache itself, but somwhere along the conversion to multi-compaonent, this feature broke. Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-10ALSA: oxygen: add some card namesClemens Ladisch1-3/+21
Instead of the generic Oxygen, use the actual card name, if known. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: do not show chip revision in card longnameClemens Ladisch2-9/+3
Apparently, the revision is 2 on all sold sound cards, so this information is not actually useful. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: X-Meridian: add S/PDIF source selectionClemens Ladisch1-2/+75
Add a mixer control to select between the on-board and extension board S/PDIF inputs for the X-Meridian (2G). Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: add digital input validity check switchClemens Ladisch1-10/+20
Add a mixer control to prevent capturing S/PDIF samples that are not marked as valid (non-audio or corrupted samples). Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: usb-audio: add Edirol SD-90 PCM supportClemens Ladisch2-3/+6
Add support for the 24-bit audio I/Os of the Edirol SD-90 interface. Reported-any-tested-by: Jim Grusendorf <alsa-user@grusendorf.ca> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: usb-audio: use enum control info helperClemens Ladisch2-16/+3
Simplify info callbacks by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: ymfpci: use enum control info helperClemens Ladisch1-9/+3
Simplify the info callback by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: cmipci: use enum control info helperClemens Ladisch1-16/+9
Simplify info callbacks by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: bt87x: use enum control info helperClemens Ladisch1-8/+2
Simplify the info callback by using the snd_ctl_enum_info() helper function. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: core, oxygen, virtuoso: add an enum control info helperClemens Ladisch7-103/+43
Introduce the helper function snd_ctl_enum_info() to fill out the elem_info fields for an enumerated control. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: add Xonar HDAV1.3 Slim supportClemens Ladisch4-16/+206
Add experimental support for the Asus Xonar HDAV1.3 Slim sound card. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: add Xonar DG supportClemens Ladisch7-5/+733
Add experimental support for the Asus Xonar DG sound card. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: add X-Meridian 2G supportClemens Ladisch2-0/+3
Add support for the AuzenTech X-Meridian 7.1 2G sound card. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: add more PCI IDsClemens Ladisch1-0/+3
Add PCI IDs for some unknown models. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: reduce MCLK in double rate modesClemens Ladisch3-7/+9
For the CSxxxx and AKxxxx DAC/ADC chips, the MCLK factor in double rate modes (64-96 kHz) can be reduced to 128x without reducing sound quality. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: simplify model-specific MCLK handlingClemens Ladisch8-95/+89
Replace the get_i2s_mclk callback with tables of MCLK values. This simplifies the MCLK-handling code in both the framework and the model- specific drivers. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: use headphone gain setting only on front DACClemens Ladisch1-0/+2
Do not apply the headphone gain offset to any but the front DAC. These DACs would not be used in headphone mode, so this saves a few register writes. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: handle DAC oversampling automaticallyClemens Ladisch1-79/+21
Remove the DAC Oversampling mixer control because this setting does not make much sense. For cards with the H6 daughterboard, 128x oversampling was disabled anyway because these high MCLK frequency would not be compatible with the connector cable. For cards without the H6 daughterboard, 128x gives a slightly higher output quality; there is no reason to reduce it to 64x except for saving power, but then these cards have not been designed to be power efficient anyway (the D2's blinkenlights cannot be disabled). Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: use lower master clock with H6 daughterboardClemens Ladisch1-13/+18
Because of the unshielded connector cable, it is important to use as low a master clock frequency as possible with the H6. For double rate modes (64-96 kHz), the MCLK rate is unconditionally lowered from 512x to 256x because the higher rate would not improve anything. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: configure correct master clock frequency on the CS2000Clemens Ladisch1-13/+12
The clock output of the CS2000, which is used as master clock for the DACs, was using half the actual master clock frequency for some reason. Using the theoretically correct frequency seems also to work in practice. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: remove non-working controls on Essence ST DeluxeClemens Ladisch1-6/+22
On the Xonar Essence ST Deluxe, remove all mixer controls that would require I2C communication with the third DAC, which does not work because of an addressing conflict with the CS2000 chip. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: change PCM1796 format to I2SClemens Ladisch1-5/+5
Change the PCM format used for the PCM1796 from left-justified to I2S to ensure that the correct format is used even for the Essence ST Deluxe's center/LFE DAC, where I2C does not work because of an address conflict with the CS2000 chip. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: wait for PCM1796 clock to become stableClemens Ladisch1-0/+5
The PCM1796 needs the master clock for I2C communication to work, so add delays after clock changes to ensure that the clock is stable when we try to write the DACs' registers. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: virtuoso: do not use fast I2C speedClemens Ladisch1-2/+2
To make the I2C communication reliable when using the H6 daughterboard, reduce the I2C clock frequency. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: fix SPI clocks slower than 6.25 MHzClemens Ladisch2-6/+6
Fix wrong register bits for SPI clock cycle times longer than 160 ns, and adjust the polling loop timeout for these speeds. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: remove oxygen_model::private_data fieldClemens Ladisch2-5/+2
The number of DACs can now be deduced from the dac_channels_mixer field, so the private_data field is no longer needed. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: oxygen: allow different number of PCM and mixer channelsClemens Ladisch7-15/+25
For cards like the Xonar HDAV1.3, differentiate between the number of PCM channels that can be played and the number of channels whose volume can be adjusted. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: hda - Add support for multiple headphone/speaker controls for RealtekTakashi Iwai1-69/+68
So far, Realtek auto-parser assumed that the multiple pins are only for line-outs, and assigned the channel names like Front, Surround, etc for the multiple outputs. But, there are devices that have multiple headphones, and these can be better controlled with the corresponding control-name like "Headphone" with indicies. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: hda - Fix multi-headphone handling for Realtek codecsTakashi Iwai1-0/+3
When multiple headphone pins are defined without line-out pins, the driver takes them as primary outputs. But it forgot to set line_out_type to HP by assuming there is some rest of HP pins. This results in some mis-handling of these pins for Realtek codec parser. It takes as if these are pure line-out jacks. Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
2011-01-10ASoC: RX1950: Enable Mic Jack during glue driver initVasily Khoruzhick1-0/+1
Enable Mic Jack during glue driver init, otherwise capture will not work. Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-10ALSA: Don't leak in sound/core/oss/pcm_oss.c::snd_pcm_hw_param_near()Jesper Juhl1-1/+3
snd_pcm_hw_param_near() will leak the memory allocated to 'save' if the call to snd_pcm_hw_param_max() returns less than zero. This patch makes sure we never leak. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: hda: Use vostro model quirk for Dell Vostro 1014Daniel T Chen1-0/+1
BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5184 A user reported on the alsa-devel mailing list that he needs to use the vostro model quirk to have audible playback, so apply it for his PCI SSID. Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com> Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xxDavid Henningsson1-8/+1
BugLink: http://bugs.launchpad.net/bugs/689036 Many new Lenovos need the ideapad quirk. Also, since the auto parser for this chip is far from optimal, the regression risk is low (although not zero). Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: HDA: Fix volume control indices for Mics (Realtek)David Henningsson1-12/+19
If more than one mic is present with different locations, e g "Front Mic" and "Rear Mic", they can use the same index (0), since their names are different. Previous behavior was to have "Front Mic" as index 1, causing it to be ignored by e g PulseAudio. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-10ALSA: HDA: Rename "Mic Boost" to "Mic Boost Volume"David Henningsson3-148/+148
BugLink: http://bugs.launchpad.net/bugs/697240 If the "Volume" suffix is not given, alsa-lib gets confused and loses the dB information at the simple element level. Boosts generally affects both playback and capture, as they are applied early in the chain. Hence no "Playback" or "Capture" in the suffix. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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