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* Merge branch 'master' into for-nextJiri Kosina2013-01-29289-3285/+7638
|\ | | | | | | | | | | | | | | Conflicts: drivers/devfreq/exynos4_bus.c Sync with Linus' tree to be able to apply patches that are against newer code (mvneta).
| * Merge remote-tracking branch 'asoc/fix/wm5100' into tmpMark Brown2013-01-101-6/+0Star
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| | * ASoC: wm5100: Remove DSP B and left justified formatsMark Brown2013-01-041-6/+0Star
| | | | | | | | | | | | | | | | | | | | | These are not supported Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | Merge remote-tracking branch 'asoc/fix/wm2200' into tmpMark Brown2013-01-101-7/+1Star
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| | * | ASoC: wm2200: Remove DSP B and left justified AIF modesMark Brown2013-01-041-6/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * | ASoC: wm2200: Fix setting dai format in wm2200_set_fmtAxel Lin2012-12-211-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | According to the defines in wm2200.h: /* * R1284 (0x504) - Audio IF 1_5 */ We should not left shift 1 bit for fmt_val when setting dai format. Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | Merge remote-tracking branch 'asoc/fix/wm2000' into tmpMark Brown2013-01-101-2/+2
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| | * | | ASoC: wm2000: Fix sense of speech clarity enableMark Brown2013-01-041-2/+2
| | | |/ | | |/| | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmpMark Brown2013-01-101-1/+22
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| | * | | ASoC: wm_adsp: Ensure that block writes are from DMA aligned addressesMark Brown2013-01-071-1/+22
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge remote-tracking branch 'asoc/fix/sta529' into tmpMark Brown2013-01-101-4/+5
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| | * | | ASoC: sta529: Fix update register bits in sta529_set_dai_fmtAxel Lin2012-12-201-4/+5
| | | |/ | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Both the mask and mode settings are wrong in current code. According to the datasheet: S2PCFG0 (0x0A) BIT[3:1] DATA_FORMAT serial interface protocol format: 000: left Justified 001: I2S (default) 010: right justified 100: PCM no delay 101: PCM delay 111: DSP Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and RIGHT_J_DATA_FORMAT. Also adds define for DATA_FORMAT_MSK. Signed-off-by: Axel Lin <axel.lin@ingics.com> Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmpMark Brown2013-01-101-2/+2
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| | * | | ASoC: sgtl5000: Fix maximum value for microphone gainFabio Estevam2012-12-241-2/+2
| | | |/ | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3. From Eric Nelson: "We also found that for the microphones we have here (commodity PC boom mics) a default value of 2 for the gain gives the best results." So change the default microphone gain as well. Signed-off-by: Eric Nelson <eric.nelson@boundarydevices.com> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge remote-tracking branch 'asoc/fix/lm49453' into tmpMark Brown2013-01-101-66/+40Star
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| | * | | ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-BMR.Swami.Reddy@ti.com2012-12-241-46/+15Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Update lm49453_reg_defs values as per LM49453 HW revision-B Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: lm49453: Fix adc, mic and sidetone volume rangesMR.Swami.Reddy@ti.com2012-12-241-19/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add adc, mic, sidetone volume ranges and appropriately added the controls. Fix the DAC HP/EP/LS/LO/HA maximum gain values. Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com> Tested-by: Vinod Koul <vinod.koul@intel.com> -- sound/soc/codecs/lm49453.c | 43 ++++++++++++++++++++++++------------------- 1 files changed, 24 insertions(+), 19 deletions(-) Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()Axel Lin2012-12-241-1/+1
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The mode variable is either 0 or 1. To update mode setting, the mask should be BIT(0) rather than BIT(1). Signed-off-by: Axel Lin <axel.lin@ingics.com> Tested-by: Omair M. Abdullah <omair.m.abdullah@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge remote-tracking branch 'asoc/fix/cs42l52' into tmpMark Brown2013-01-101-3/+1Star
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| | * | | ASoC: cs42l52: Catch no-match case in cs42l52_get_clkAxel Lin2012-12-241-3/+1Star
| | |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the case of no-match, return -EINVAL instead of 0. Since we assign i to ret in the for loop, ret always less than ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret. Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | Merge remote-tracking branch 'asoc/fix/cs4271' into tmpMark Brown2013-01-101-3/+3
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| | * | | ASoC: cs4271: fix property checkDaniel Mack2012-12-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The driver had the property check for 'cirrus,amutec_eq_bmutec' the wrong way around. That happens if you misspell the property in the bindings during tests. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: cs4271: fix sparse warningDaniel Mack2012-12-021-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Make the flag in the pdata of type bool to fix a sparse warning. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | Merge remote-tracking branch 'asoc/fix/core' into tmpMark Brown2013-01-102-3/+33
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| | * | | | ASoC: core: fix the memory leak in case of remove_aux_dev()Chuansheng Liu2012-12-271-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When probing aux_dev, initializing is as below: device_initialize() device_add() So when remove aux_dev, we need do as below: device_del() device_put() Otherwise, the rtd_release() will not be called. So here using device_unregister() to replace device_del(), like the action in soc_remove_link_dais(). Signed-off-by: liu chuansheng <chuansheng.liu@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: core: fix the memory leak in case of device_add() failureChuansheng Liu2012-12-271-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After called device_initialize(), even device_add() returns error, we still need use the put_device() to release the reference to call rtd_release(), which will do the free() action. Signed-off-by: liu chuansheng <chuansheng.liu@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: soc-core: Remove unused 'ret' variableFabio Estevam2012-12-241-1/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced the following build warning: sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable] Remove the unused 'ret' variable. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: core: Fix SOC_DOUBLE_RANGE() macrosMark Brown2012-12-201-2/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
| | * | | | ASoC: pcm: allow backend hardware to be freed in pause statePatrick Lai2012-12-201-0/+1
| | | |_|/ | | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai@codeaurora.org> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | ASoC: arizona: Remove DSP B and left justified AIF modesMark Brown2013-01-041-6/+0Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | ASoC: wm5102: Improve speaker enable performanceMark Brown2013-01-021-2/+46
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: arizona: Correct FLL source definitionsMark Brown2012-12-241-9/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The FLL source constants were numbered as a simple enumeration but were being used in the code as direct values to be written to the registers. Renumber the constants to reflect the usage. Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | ASoC: arizona: Do proper shift for setting AIF rateAxel Lin2012-12-241-1/+2
| | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */ Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate. Signed-off-by: Axel Lin <axel.lin@ingics.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | Merge tag 'sound-3.8' of ↵Linus Torvalds2012-12-206-57/+108
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "This update contains overall only driver-specific fixes. Slightly large LOC are seen in usb-audio driver for a couple of new device quirks and cs42l71 ASoC driver for enhanced features. The others are a few small (regression) fixes HD-audio, and yet other small / trival ASoC fixes." * tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card: ALSA: HDA: Fix sound resume hang ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs ASoC: atmel-ssc: change disable to disable in dts node ASoC: Prevent pop_wait overwrite ALSA: usb-audio: ignore-quirk for HP Wireless Audio ALSA: hda - Always turn on pins for HDMI/DP ALSA: hda - Fix pin configuration of HP Pavilion dv7 ASoC: core: Fix splitting of log messages ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT ASoC: cs42l73: Add DAPM events for power down. ASoC: cs42l73: Add DMIC's as DAPM inputs. ASoC: sigmadsp: Fix endianness conversion issue ASoC: tpa6130a2: Use devm_* APIs
| | * | Merge remote-tracking branch 'asoc/topic/tpa6130a2' into asoc-nextMark Brown2012-12-151-17/+6Star
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| | | * | ASoC: tpa6130a2: Use devm_* APIsSachin Kamat2012-12-091-17/+6Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Converted to use devm_gpio_request and devm_regulator_get APIs. These are device managed and make error handling and cleanup a bit simpler. Cc: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge remote-tracking branch 'asoc/topic/log' into asoc-nextMark Brown2012-12-151-5/+5
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| | | * | | ASoC: core: Fix splitting of log messagesMark Brown2012-12-101-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Don't wrap log messages over multiple lines, it makes them hard to grep for. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-nextMark Brown2012-12-151-27/+89
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| | | * | | | ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUTPaul Handrigan2012-12-101-19/+13Star
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since VSP only has one power bit, only provide one DAPM widget. Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: cs42l73: Add DAPM events for power down.Paul Handrigan2012-12-101-8/+72
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add power down delays between setting PDN and MCLKDIS for spk amp, spklo amp, and ear amp. Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: cs42l73: Add DMIC's as DAPM inputs.Paul Handrigan2012-12-101-0/+4
| | | | |/ / | | | |/| | | | | | | | | | | | | | | | | | | | Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge remote-tracking branch 'asoc/topic/core' into asoc-nextMark Brown2012-12-152-7/+7
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| | | * | | | ASoC: Prevent pop_wait overwriteMisael Lopez Cruz2012-12-152-7/+7
| | | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge remote-tracking branch 'asoc/fix/sigmadsp' into asoc-nextMark Brown2012-12-151-1/+1
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| | | * | | | ASoC: sigmadsp: Fix endianness conversion issueLars-Peter Clausen2012-12-101-1/+1
| | | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The 'addr' field of the sigma_action struct is stored as big endian in the firmware file. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | | Merge tag 'sound-3.8' of ↵Linus Torvalds2012-12-13281-3013/+6353
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This update contains a fairly wide range of changes all over in sound subdirectory, mainly because of UAPI header moves by David and __dev* annotation removals by Bill. Other highlights are: - Introduced the support for wallclock timestamps in ALSA PCM core - Add the poll loop implementation for HD-audio jack detection - Yet more VGA-switcheroo fixes for HD-audio - New VIA HD-audio codec support - More fixes on resource management in USB audio and MIDI drivers - More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite, Roland VG-99, etc - Add support for FastTrack C400 usb-audio - Clean ups in many drivers regarding firmware loading - Add PSC724 Ultiimate Edge support to ice1712 - A few hdspm driver updates - New Stanton SCS.1d/1m FireWire driver - Standardisation of the logging in ASoC codes - DT and dmaengine support for ASoC Atmel - Support for Wolfson ADSP cores - New drivers for Freescale/iVeia P1022 and Maxim MAX98090 - Lots of other ASoC driver fixes and developments" Fix up trivial conflicts. And go out on a limb and assume the dts file 'status' field of one of the conflicting things was supposed to be "disabled", not "disable" like in pretty much all other cases. * tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits) ALSA: hda - Move runtime PM check to runtime_idle callback ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522 ALSA: hda - Avoid doubly suspend after vga switcheroo ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3 ALSA: hda - Check validity of CORB/RIRB WP reads ALSA: hda - use usleep_range in link reset and change timeout check ALSA: HDA: VIA: Add support for codec VT1808. ALSA: HDA: VIA Add support for codec VT1705CF. ASoC: codecs: remove __dev* attributes ASoC: utils: remove __dev* attributes ASoC: ux500: remove __dev* attributes ASoC: txx9: remove __dev* attributes ASoC: tegra: remove __dev* attributes ASoC: spear: remove __dev* attributes ASoC: sh: remove __dev* attributes ASoC: s6000: remove __dev* attributes ASoC: OMAP: remove __dev* attributes ASoC: nuc900: remove __dev* attributes ASoC: mxs: remove __dev* attributes ASoC: kirkwood: remove __dev* attributes ...
| | * | | | ASoC: codecs: remove __dev* attributesBill Pemberton2012-12-09101-412/+412
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | CONFIG_HOTPLUG is going away as an option. As result the __dev* markings will be going away. Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, and __devexit. Signed-off-by: Bill Pemberton <wfp5p@virginia.edu> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: utils: remove __dev* attributesBill Pemberton2012-12-091-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | CONFIG_HOTPLUG is going away as an option. As result the __dev* markings will be going away. Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, and __devexit. Signed-off-by: Bill Pemberton <wfp5p@virginia.edu> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: ux500: remove __dev* attributesBill Pemberton2012-12-093-9/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | CONFIG_HOTPLUG is going away as an option. As result the __dev* markings will be going away. Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst, and __devexit. Signed-off-by: Bill Pemberton <wfp5p@virginia.edu> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
generated by cgit v1.2.3-55-g7522 (git 2.39.0) at 2024-07-12 00:28:27 +0200