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commit 2e7ee15ced914e109a1a5b6dfcd463d846a13bd5 upstream.
Also fix return values for headphone switch updates.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cb6f66a2d278e57a6c9d8fb59bd9ebd8ab3965c2 upstream.
The registers of max98088 are 8 bits, not 16 bits. This bug causes the
contents of registers to be overwritten with bad values when the codec
is suspended and then resumed.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 016fcab8ff46fca29375d484226ec91932aa4a07 upstream.
According to the sgtl5000 reference manual, the default value of CHIP_SSS_CTRL
is 0x10.
Reported-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 5c78dfe87ea04b501ee000a7f03b9432ac9d008c upstream.
SGTL5000_PLL_FRAC_DIV_MASK is used to mask bits 0-10 (11 bits in total) of
register CHIP_PLL_CTRL, so fix the mask to accomodate all this bit range.
Reported-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When calling snd_soc_dapm_sync(), it eventually tries to lock the same mutex
already locked in snd_soc_dapm_put_volsw_aic3x() and a deadlock occurs. By
moving the mutex unlock to just before snd_soc_dapm_sync(), this deadlock is
prevented. This problem was introduced in Linux 3.5
Signed-off-by: Andreas Irestål <Andreas.Irestal@axis.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Signed-off-by: Nicolas Schichan <nschichan@freebox.fr>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Nicolas Schichan <nschichan@freebox.fr>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The array 'drc_cfg' of size 3 may use index value -22 (EINVAL)
The array 'retune_mobile_cfg' of size 3 may use index value -22 (EINVAL)
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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During recent refactoring the code to report removal when MICDET reports
an absent microphone was removed, causing problems for systems which rely
solely on the MICDET for this functionality. Restore it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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request_threaded_irq() rejects calls which both do not specify a handler
(indicating that the primary IRQ handler should be used) and do not set
IRQF_ONESHOT because the combination is unsafe with level-triggered
interrupts. It is safe in this case, though, since max98090 IRQs are
edge-triggered and the interrupts aren't ACK'ed until the codec's IRQ
status register is read. Because of this, an IRQF_ONESHOT interrupt
doesn't really make a difference, but request one anyway in order to make
request_threaded_irq() happy.
Signed-off-by: Andrew Bresticker <abrestic@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The default register value for MASTERA_VOL is 0x00, the same as
MASTERB_VOL.
Signed-off-by: Nicolas Schichan <nschichan@freebox.fr>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The mask should define the bits to change in the register, not the
bits to preserve.
This fixes the inadvertent changes of the "Headphone Analog Gain"
value during mute/unmute.
Signed-off-by: Nicolas Schichan <nschichan@freebox.fr>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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controls.
Signed-off-by: Nicolas Schichan <nschichan@freebox.fr>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Nicolas Schichan <nschichan@freebox.fr>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Add callback to initialise the speaker in the core following the recent
changes to handling of integration with the thermal interrupts.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Fix to return -ENOMEM in the memory malloc of 'out' and 'img_swap' error
handling case instead of 0, as done elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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AD slots definitions for ab8500 codec were erroneously swapped between
even and odd channels. Fix this by swapping the definitions to be
coherent with the channel number.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When set dmic_samplephase and dmic_clk_rate bits for dmic_cfg,
current code checks pdata->dmic_data_sel which is wrong.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
Revert "ALSA: hda - Don't set up active streams twice"
ALSA: Add comment for control TLV API
ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
ALSA: HDA: Fix Oops caused by dereference NULL pointer
ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
ALSA: mips/hal2: Remove redundant platform_set_drvdata()
ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
sound: Fix make allmodconfig on MIPS
ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers
ALSA: atmel: Remove redundant platform_set_drvdata()
ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
ASoC: wm8994: missing break in wm8994_aif3_hw_params()
ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A few more bug fixes, the DAPM clock fix is actually a driver specific
one since currently there's only one user of the clock support due to
the problems relying on the clock API.
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The missing break here means that we always return early and the
function is a no-op.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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The hardware revision of the codec is based at 0x40. Subtract that
before convering to ASCII. The same as it is done for 98095.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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There's already a device revision stored in the core data structure,
don't duplicate it in the CODEC driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This helps to ensure a smooth startup when we restore.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Some systems use the audio CODEC to clock a DAI with multiple data lines
in parallel, meaning that bit clocks are only required for a smaller number
of channels than data is sent for. In some cases providing the extra bit
clocks can take the other devices on the audio bus out of spec.
Support such systems by allowing a maximum number of channels to be
specified.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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