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2012-07-17ALSA: opti9xx: Fix section mismatch by PM supportTakashi Iwai1-1/+1
In the previous commit, snd_opti9xx_configure() is called from the resume handler but it's still marked as __devinit. Fix it. Reported-by: Fengguang Wu <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-17ALSA: snd-opti9xx: Implement suspend/resumeOndrej Zary1-4/+63
Implement suspend/resume support for Opti 92x and 93x chips. Tested with Opti 929A+AD1848 and Opti 931. Signed-off-by: Ondrej Zary <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-17ALSA: hda - Add new GPU codec ID to snd-hdaAaron Plattner1-0/+2
Vendor ID 0x10de0051 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <[email protected]> Acked-by: Andy Ritger <[email protected]> Reviewed-by: Daniel Dadap <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: hda - Fix driver type of Haswell controller to AZX_DRIVER_SCHTakashi Iwai1-1/+1
According to Xingchao, This works for HDMI audio, otherwise there's blocking issue. Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: hda - add Haswell HDMI codec idWang Xingchao1-0/+2
0x80862807 is HDMI id for Haswell HDA. Signed-off-by: Wang Xingchao <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: hda - Add DeviceID for Haswell HDAWang Xingchao1-0/+5
this patch add proper id for Haswell HDA Controller. [Added AZX_DCAPS_POSFIX_COMBO flag by tiwai] Signed-off-by: Wang Xingchao <[email protected]> Acked-by: Jaroslav Kysela <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: wss_lib: Fix resume on Yamaha OPL3-SAxOndrej Zary1-0/+4
Yamaha OPL3-SAx chips don't resume properly when playback is running - garbage is played after resume. Restoring the CS4231_PLAYBK_FORMAT register last fixes the problem. Signed-off-by: Ondrej Zary <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: wss_lib: fix suspend/resumeOndrej Zary1-1/+0
By setting SNDRV_PCM_INFO_RESUME, wss_lib claims that it can restore the card state fully on resume. But in fact, it can't as DMA is not restored so any playback/capture running during suspend will fail to continue after resume. Remove SNDRV_PCM_INFO_RESUME flag from pcm info field to fix the problem. Signed-off-by: Ondrej Zary <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: es1938: replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGEClemens Ladisch1-15/+10
Instead of the hard-to-mantain TLV_DB_RANGE_HEAD macro, use DECLARE_TLV_DB_RANGE, which computes its size automatically. (Also make this data const on the way.) Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: tlv: add DECLARE_TLV_DB_RANGE()Clemens Ladisch1-0/+4
Add a DECLARE_TLV_DB_RANGE() macro so that dB range information can be specified without having to count the items manually for TLV_DB_RANGE_HEAD(). Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: tlv: add DECLARE_TLV_CONTAINER()Clemens Ladisch1-0/+5
Add the DECLARE_TLV_CONTAINER() macro to allow having static TLVs containing more than one item. Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-16ALSA: tlv: compute TLV_*_ITEM lengths automaticallyClemens Ladisch1-8/+12
Add helper macros with a little bit of preprocessor magic to automatically compute the length of a TLV item. This lets us avoid having to compute this by hand, and will allow to use items that do not use a fixed length. Signed-off-by: Clemens Ladisch <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-15ASoC: Convert S3C2412 I2S driver to gpiolib APISylwester Nawrocki1-7/+3
The s3c2410_gpio* calls are obsolete and have been scheduled for removal since several kernel releases. Remove them and use common gpiolib API. This patch is a prerequisite for removal of the obsolete S3C24XX SoC GPIO definitions. Compile tested only. Cc: Ben Dooks <[email protected]> Signed-off-by: Sylwester Nawrocki <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-15ASoC: Convert S3C24XX I2S driver to gpiolib APISylwester Nawrocki1-7/+3
The s3c2410_gpio* calls are obsolete and have been scheduled for removal since several kernel releases. Remove them and use common gpiolib API. This patch is a prerequisite for removal of the obsolete S3C24XX SoC GPIO definitions. Tested on Micro2440-SDK. Cc: Ben Dooks <[email protected]> Signed-off-by: Sylwester Nawrocki <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-13ASoC: dapm: Fix compilation warningMarek Belisko1-0/+1
Fix following: sound/soc/soc-dapm.c: In function ‘dapm_clock_event’: sound/soc/soc-dapm.c:1021:1: warning: control reaches end of non-void function [-Wreturn-type] Signed-off-by: Marek Belisko <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-13ALSA: snd-usb: move calls to usb_set_interfaceDaniel Mack2-89/+41
The rework of the snd-usb endpoint logic moved the calls to snd_usb_set_interface() into the snd_usb_endpoint implemenation. This changed the order in which these calls are issued to the device, and thereby caused regressions for some webcams. Fix this by moving the calls back to pcm.c for now to make it work again and use snd_usb_endpoint_activate() to really tear down all remaining URBs in the flight, consequently fixing another regression caused by USB packets on the wire after altsetting 0 has been selected. Signed-off-by: Daniel Mack <[email protected]> Reported-and-tested-by: Philipp Dreimann <[email protected]> Reported-by: Joseph Salisbury <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-12ASoC: wm8962: Redo early init of the part on resumeMark Brown1-0/+3
Ensure robust startup of the part by going through the reset procedure prior to resyncing the full register cache, avoiding potential intermittent faults in some designs. Signed-off-by: Mark Brown <[email protected]> Cc: [email protected]
2012-07-11ASoC: Free memory in the error paths of soc_of_parse_audio_routing()Matthias Kaehlcke1-0/+2
Release the memory of the routing table before leaving the function upon errors in the device tree Signed-off-by: Matthias Kaehlcke <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-11ASoC: wm5110: Add audio CODEC driverMark Brown4-0/+979
The WM5110 is a highly integrated low power audio subsystem for smartphones, tablets and other portable audio devices. It combines an advanced DSP feature set with a flexible, high performance audio hub CODEC. This patch adds the audio CODEC driver for the device. Signed-off-by: Mark Brown <[email protected]>
2012-07-11ASoC: STA529: fix an error messageDan Carpenter1-1/+2
GCC complains that "ret" is uninitialized here. Signed-off-by: Dan Carpenter <[email protected]> Acked-By: Rajeev Kumar <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-10ASoC: arizona: Support variable FLL VCO multipliersMark Brown3-3/+7
Some Arizona chips have a higher frequency for the FLL VCO, support this in the common code. Signed-off-by: Mark Brown <[email protected]>
2012-07-10ASoC: tlv320aic3x: add input clock selectionJiri Prchal2-0/+14
This patch adds input selection of main codec clock - from what pin. Both registers set same value since codec uses clock divider or pll at one time. Signed-off-by: Jiri Prchal <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-10ASoC: tlv320aic3x: add AGC settingsJiri Prchal1-0/+25
This patch adds AGC target level and times settings for TLV320AIC3x. Enums uses small arrays of two channels left and right since it uses different registers. Signed-off-by: Jiri Prchal <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-09ASoC: tlv320aic3x: add deemphasis switchJiri Prchal1-0/+3
This patch adds missing deemphasis switch. Signed-off-by: Jiri Prchal <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-09ASoC: wm5102: Fix cut'n'paste for digital volume registersMark Brown1-3/+3
The analogue PGA shifts were used; this makes no practical difference as the values are the same. Signed-off-by: Mark Brown <[email protected]>
2012-07-09ASoC: arizona: Add IN4 to the mixer tablesMark Brown2-1/+5
Some devices have four input structures rather than three. Signed-off-by: Mark Brown <[email protected]>
2012-07-09ASoC: arizona: Export dai_opsMark Brown1-0/+1
Signed-off-by: Mark Brown <[email protected]>
2012-07-06ASoC: omap-mcpdm: Add missing MODULE_ALIASPeter Ujfalusi1-0/+1
The MODULE_ALIAS() was missing from the driver. Signed-off-by: Peter Ujfalusi <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-06ASoC: arizona: Change DAPM routes for AIF clocks when we change themMark Brown1-3/+29
Signed-off-by: Mark Brown <[email protected]> Acked-by: Liam Girdwood <[email protected]>
2012-07-06ASoC: dapm: Allow routes to be deleted at runtimeMark Brown2-0/+79
Since we're now relying on DAPM for things like enabling clocks when we reparent the clocks for widgets we need to either use conditional routes (which are expensive) or remove routes at runtime. Add a route removal API to support this use case. Signed-off-by: Mark Brown <[email protected]> Acked-by: Liam Girdwood <[email protected]>
2012-07-06ASoC: dapm: Mark widgets as dirty when a route is addedMark Brown1-0/+4
If we add a new route at runtime then we'll need to recheck the connections to the affected widgets. Signed-off-by: Mark Brown <[email protected]> Acked-by: Liam Girdwood <[email protected]>
2012-07-06ASoC: dpcm: Allow FE to be opened without valid BE routes.Liam Girdwood1-3/+1
Some userspace will open a PCM device and then configure mixers for routing before triggering. This patch allows userspace to do this sequence. Signed-off-by: Liam Girdwood <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-06ASoC: twl6040: fix spelling mistakeSimon Wilson1-1/+1
Fix spelling mistake in "High-Performance" option of twl6040 power mode. Signed-off-by: Simon Wilson <[email protected]> Signed-off-by: Liam Girdwood <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-06ASoC: dapm: Make sure all dapm contexts are updatedLiam Girdwood1-3/+3
Make sure we set the bias level for all DAPM contexts when changing level. Signed-off-by: Liam Girdwood <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-06ASoC: pcm: Clean up logging in soc_new_pcm()Liam Girdwood1-4/+4
Use dev_ style logging throughout soc_new_pcm() Signed-off-by: Liam Girdwood <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-06ASoC: dapm: Fix locking during codec shutdownLiam Girdwood1-0/+5
Codec shutdown performs a DAPM power sequence that might cause conflicts and/or race conditions if another stream power event is running simultaneously. Use card's dapm mutex to protect any potential race condition between them. Signed-off-by: Misael Lopez Cruz <[email protected]> Signed-off-by: Liam Girdwood <[email protected]> Signed-off-by: Mark Brown <[email protected]> Cc: [email protected]
2012-07-06ALSA: usb-audio: Fix the first PCM interface assignmentTakashi Iwai1-2/+2
In the new PCM streaming logic, the interface number is assigned to usb stream instance (subs->interface) after the format and rate setups are succeeded, but some codes are still passing subs->interface as the reference to helper functions. This leads to initializing with an invalid iface number (-1). This patch replaces the wrong references with the ones from the target fmt correctly. Signed-off-by: Takashi Iwai <[email protected]>
2012-07-05ASoC: wm1250-ev1: Flag all supported rates in the DAIMark Brown1-2/+5
Not previously noticed due to normal usage being with CODEC<->CODEC links. Signed-off-by: Mark Brown <[email protected]>
2012-07-05ASoC: wm5102: Allow routing through the ASRCsMark Brown1-1/+29
This enables common telephony use cases. Signed-off-by: Mark Brown <[email protected]>
2012-07-05ASoC: arizona: Enable ASYNCCLK domain for audio interfacesMark Brown1-6/+24
If an audio interface is configured to use ASYNCCLK then update the asynchronous sample rate rather than one of our primary sample rates. Signed-off-by: Mark Brown <[email protected]>
2012-07-05ASoC: imx-mc13783: Add audmux settings for mx27pdkFabio Estevam1-16/+33
mx27pdk board also has a mc13783 codec. Add support for it and do a run-time machine type check to perform the correct audiomux settings. Signed-off-by: Fabio Estevam <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-05ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000Hui Wang1-2/+1
If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR register (means Tx clock pin is input), we don't need to set IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL settings are ignored". Signed-off-by: Hui Wang <[email protected]> Acked-by: Shawn Guo <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-05ALSA: pcm: Make constraints lists constMark Brown2-3/+3
They aren't modified by the core so the drivers can declare them const. Signed-off-by: Mark Brown <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-05ALSA: hda - Always call standard unsolicited event for Realtek codecsDavid Henningsson1-18/+3
With the model parsers out of the way, we have no custom unsol events to worry about, we can therefore simplify the code. In addition, this fixes a bug on the ASUS TC710, which has only a headphone jack and a mic jack, but no internal mic or speakers. Therefore the unsol_event pointer was not set, and as a result, the jack kcontrols were not correctly updated. BugLink: https://bugs.launchpad.net/bugs/1021192 Signed-off-by: David Henningsson <[email protected]> Signed-off-by: Takashi Iwai <[email protected]>
2012-07-04ASoC: arizona: Be more forgiving in BCLK selectionMark Brown1-1/+2
Allow any BCLK which can be divided down to generate LRCLK, not just the lowest possible BCLK to clock out the samples. Signed-off-by: Mark Brown <[email protected]>
2012-07-04ASoC: arizona: Implement AIF clock configurationMark Brown3-0/+118
Allow the user to select which of the system clocks each AIF is referenced to and constran the DAI to the set of frequencies which can be generated from that clock. Signed-off-by: Mark Brown <[email protected]>
2012-07-04ASoC: arizona: Rename current rates tables to bclk_ratesMark Brown1-6/+6
They're the rates for the BCLK, not for the sample rate, so rename so that we don't confuse ourselves. Signed-off-by: Mark Brown <[email protected]>
2012-07-04ASoC: dwc: Staticise non-exported i2s_start()Mark Brown1-1/+2
Signed-off-by: Mark Brown <[email protected]>
2012-07-04ASoC: SPEAr spdif_out: Add spdif out supportVipin Kumar2-0/+468
This patch implements the spdif out driver for ST peripheral. This peripheral implements IEC60958 standard Signed-off-by: Vipin Kumar <[email protected]> Signed-off-by: Rajeev Kumar <[email protected]> Signed-off-by: Mark Brown <[email protected]>
2012-07-04ASoC: STA529: Add support for STA529 Audio CodecRajeev Kumar3-0/+447
The STA529 is a digital stereo class-D audio amplifier. It includes an audio DSP, an ST proprietary high-efficiency class-D driver and CMOS power output stage. It is intended for high-efficiency digital-to-power-audio conversion for portable applications. Signed-off-by: Rajeev Kumar <[email protected]> Signed-off-by: Mark Brown <[email protected]>
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