We should not copy the return value into this val since it's supposed to
get the value of the register not the success result of regmap_read().
Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit c9e065c27f ("ASoC: dapm: Make sure to always update the DAPM graph
in _put_volsw()") stopped updating register values in those cases where
initial after boot state of kcontrol appears to not change but where
register value still needs update because it is not in sync with the
kcontrol state.
Fix this by doing snd_soc_test_bits() unconditionally as it was before but
by using separate flags for kcontrol and register state changes. This allow
both DAPM graph to be updated when disabling auto-muted control and update
register if it is out-of-sync in respect of kcontrol state.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 7189eb9b8f.
It will use LPIB to get the DMA position on Broadwell HDMI Audio.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Broadwell HDMI can't use position buffer reliably, force to use LPIB
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the SigmaDSP module is built-in, but the I2C core is build as a module
we'll get a undefined reference:
sound/built-in.o: In function `sigma_action_write_i2c':
:(.text+0x5d8d4): undefined reference to `i2c_master_send'
This can happen if a audio driver that is using the regmap SigmaDSP interface is
built into the kernel, but core I2C support is build as a module. To fix this
split the SigmaDSP module into three modules, one module providing the core
infrastructure and two small modules implementing the regmap and I2C interfaces.
This allows e.g. the core infrastructure and regmap support to be built into the
kernel while I2C support can still be build as a module.
Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
These two new pin tables can fix headset mic problems for several
new Dell machines.
And also delete some machines from old quirk table since the existing
pin talbes already cover them.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From e7a94bb7fb871c73cc85712d89c1f48d0271c1be Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 5 Jun 2014 12:31:28 +0200
Subject: [PATCH] ASoC: MMP audio needs sram support
Building the pxa/mmp audio driver without support for the mmp
sram driver enabled results in this link error:
sound/built-in.o: In function `mmp_pcm_free_dma_buffers':
:(.text+0x3e734): undefined reference to `sram_get_gpool'
sound/built-in.o: In function `mmp_pcm_new':
:(.text+0x3e7c0): undefined reference to `sram_get_gpool'
The sram driver is cannot be manually enabled and needs to
be turned on by selecting MMP_SRAM from each module that
needs it, which is what this patch does.
Ideally, MMP should move over to the generic SRAM support, but
for the moment, we can avoid the build error.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
New codec support for ALC891.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781,
and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for incoming
isochronous stream and duplex streams with timestamp synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc
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Merge tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next
Pull sound updates from Takashi Iwai:
"At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC:
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with
simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651
and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for
incoming isochronous stream and duplex streams with timestamp
synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc"
* tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits)
ASoC: Fix wrong argument for card remove callbacks
ASoC: free jack GPIOs before the sound card is freed
ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation
ASoC: cache: Fix error code when not using ASoC level cache
ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
ASoC: add RT5677 CODEC driver
ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
ASoC: Add helper functions to cast from DAPM context to CODEC/platform
ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
ASoC: wm9713: correct mono out PGA sources
ALSA: synth: emux: soundfont.c: Cleaning up memory leak
ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
ASoC: fsl-ssi: Use regmap
ASoC: fsl-ssi: reorder and document fsl_ssi_private
...
Sometimes PORT_EXIT messages are lost when a process is exiting.
This happens if you subscribe to the announce port with client A,
then subscribe to the announce port with client B, then kill client A.
Client B will not see the PORT_EXIT message because client A's port is
closing and is earlier in the announce port subscription list. The
for each loop will try to send the announcement to client A and fail,
then will stop trying to broadcast to other ports. Killing B works fine
since the announcement will already have gone to A. The CLIENT_EXIT
message does not get lost.
How to reproduce problem:
*** termA
$ aseqdump -p 0:1
0:1 Port subscribed 0:1 -> 128:0
*** termB
$ aseqdump -p 0:1
*** termA
0:1 Client start client 129
0:1 Port start 129:0
0:1 Port subscribed 0:1 -> 129:0
*** termB
0:1 Port subscribed 0:1 -> 129:0
*** termA
^C
*** termB
0:1 Client exit client 128
<--- expected Port exit as well (before client exit)
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently mutex_destroy() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
[fixed a typo in changelog by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type.
But this member is 1 byte. Although the value is between 0x00-0xff, a constant
has 0x10000. This constant is meaningless.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To reverse a pointer for the ring buffer, subtraction by buffer
size is better than assignment to the beginning of the buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All assignment for local variables in these functions are not related to
critical section.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_event_dup returns -ENOMEM in some buffer-full conditions,
but usually returns -EAGAIN. Make -EAGAIN trigger the overflow
condition in snd_seq_fifo_event_in so that the fifo is cleared
and -ENOSPC is returned to userspace as stated in the alsa-lib docs.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have in the past added 'depends on I2C' for some of the PXA boards
after hitting randconfig build bugs. I have seens a couple of new
bugs in this area during the linux-next cycle for 3.16, after it
became possible to build some more PXA machines with I2C disabled.
To shut this up for good, this adds the dependency to every board
that uses I2C as the interface to the codec. I have gone through
all board files and verified that they all either use AC97 or
I2C, and this annotates the latter. Some of these already enable
I2C from mach-pxa/Kconfig, but since that can change it's better
to be explicit here.
The link error that can result otherwise happens when CONFIG_I2C
is set to 'm' and the codec driver is built-in as a result of being
selected by the platform specific glue.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is
freed] introduced snd_soc_card remove callbacks to a few drivers, but
they are implemented with a wrong argument type. The callback should
receive snd_soc_card pointer instead of snd_soc_pcm_runtime.
Fixes: e1d4d3c854 ('ASoC: free jack GPIOs before the sound card is freed')
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
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Merge tag 'asoc-v3.16-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.16
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
This is the same change as commit fb6b8e7144 "ASoC: tegra: free jack
GPIOs before the sound card is freed", but applied to all other ASoC
machine drivers where code inspection indicates the same problem exists.
That commit's description is:
==========
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, guard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
==========
Note that I have not even compile-tested this in most cases, since most
of the drivers rely on specific mach-* support I don't have enabled, and
don't support COMPILE_TEST. Testing by the relevant board maintainers
would be useful.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The comment for fcp_avc_transaction() describes it doesn't support this type
of operation. But it was already supported by this commit.
00a7bb81c2
ALSA: firewire-lib: Add support for deferred transaction
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A quite large set of SoC updates this cycle. In no particular order:
- Multi-cluster power management for Samsung Exynos, adding support for
big.LITTLE CPU switching on EXYNOS5
- SMP support for Marvell Armada 375 and 38x
- SMP rework on Allwinner A31
- Xilinx Zynq support for SOC_BUS, big endian
- Marvell orion5x platform cleanup, modernizing the implementation and
moving to DT.
- _Finally_ moving Samsung Exynos over to support MULTIPLATFORM, so
that their platform can be enabled in the same kernel binary as most
of the other v7 platforms in the tree. \o/ The work isn't quite complete,
there's some driver fixes still needed, but the basics now work.
New SoC support added:
- Freescale i.MX6SX
- LSI Axxia AXM55xx SoCs
- Samsung EXYNOS 3250, 5260, 5410, 5420 and 5800
- STi STIH407
Plus a large set of various smaller updates for different platforms. I'm
probably missing some important one here.
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Merge tag 'soc-for-3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc into next
Pull part one of ARM SoC updates from Olof Johansson:
"A quite large set of SoC updates this cycle. In no particular order:
- Multi-cluster power management for Samsung Exynos, adding support
for big.LITTLE CPU switching on EXYNOS5
- SMP support for Marvell Armada 375 and 38x
- SMP rework on Allwinner A31
- Xilinx Zynq support for SOC_BUS, big endian
- Marvell orion5x platform cleanup, modernizing the implementation
and moving to DT.
- _Finally_ moving Samsung Exynos over to support MULTIPLATFORM, so
that their platform can be enabled in the same kernel binary as
most of the other v7 platforms in the tree. \o/
The work isn't quite complete, there's some driver fixes still
needed, but the basics now work.
New SoC support added:
- Freescale i.MX6SX
- LSI Axxia AXM55xx SoCs
- Samsung EXYNOS 3250, 5260, 5410, 5420 and 5800
- STi STIH407
plus a large set of various smaller updates for different platforms.
I'm probably missing some important one here"
* tag 'soc-for-3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (281 commits)
ARM: exynos: don't run exynos4 l2x0 setup on other platforms
ARM: exynos: Fix "allmodconfig" build errors in mcpm and hotplug
ARM: EXYNOS: mcpm rename the power_down_finish
ARM: EXYNOS: Enable mcpm for dual-cluster exynos5800 SoC
ARM: EXYNOS: Enable multi-platform build support
ARM: EXYNOS: Consolidate Kconfig entries
ARM: EXYNOS: Add support for EXYNOS5410 SoC
ARM: EXYNOS: Support secondary CPU boot of Exynos3250
ARM: EXYNOS: Add Exynos3250 SoC ID
ARM: EXYNOS: Add 5800 SoC support
ARM: EXYNOS: initial board support for exynos5260 SoC
clk: exynos5410: register clocks using common clock framework
ARM: debug: qcom: add UART addresses to Kconfig help for APQ8084
ARM: sunxi: allow building without reset controller
Documentation: devicetree: arm: sort enable-method entries
ARM: rockchip: convert smp bringup to CPU_METHOD_OF_DECLARE
clk: exynos5250: Add missing sysmmu clocks for DISP and ISP blocks
ARM: dts: axxia: Add reset controller
power: reset: Add Axxia system reset driver
ARM: axxia: Adding defconfig for AXM55xx
...
It is not an error to have no cache so we shouldn't return an error code
and cause our callers to fail, just silently do nothing instead. Thanks
to Jarkko for identify the problematic commit.
Reported-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Reported-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The conversion to a fixup table for Replacer model with ALC260 in
commit 20f7d928 took the wrong widget NID for COEF setups. Namely,
NID 0x1a should have been used instead of NID 0x20, which is the
common node for all Realtek codecs but ALC260.
Fixes: 20f7d928fa ('ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser')
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correcion of wrong fixup entries add in commit ca8f0424 to replace
static model quirk for PB V7900 laptop (will model).
[note: the removal of ALC260_FIXUP_HP_PIN_0F chain is also needed as a
part of the fix; otherwise the pin is set up wrongly as a headphone,
and user-space (PulseAudio) may be wrongly trying to detect the jack
state -- tiwai]
Fixes: ca8f04247e ('ALSA: hda/realtek - Add the fixup codes for ALC260 model=will')
Signed-off-by: Ronan Marquet <ronan.marquet@orange.fr>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio
data means Valid Length Code (VBL). Ths value is:
- b00 for 24 bits sample (label is 0x40)
- b01 for 20 bits sample (label is 0x41)
- b10 for 16 bits sample (label is 0x42)
But current firewire-lib apply 24 bits label for both of 16/24 bits samples.
As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them
have a behaviour to ignore the label. They can generate correct sound even
if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only
for Raw Audio data channel, but also for IEC 60958 Conformant data channel.
So there is little possibility of regression.
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the function "get_clk_info" to RL6231 shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the function of the PLL clock calculation to RL6231 shared
support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the RL6231 class device shared support for RT5640, RT5645 and
RT5651. The function of the DMIC clock calculation can be shared by RL6231
shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we cannot make sure the 'reg_size' will always be none zero here,
and then if 'reg_size' equals to zero, the kzalloc() will return ZERO_SIZE_PTR,
which equals to ((void *)16).
So this patch fix this with just doing the 'reg_size' zero check before calling
kzalloc().
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ARRAY_SIZE() was intended here instead of sizeof(). The
"bridgeco_freq_table" array holds integers so the original condition is
never true.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mono output PGA input only has four possible sources, so
omit the rest.
Signed-off-by: Matt Reimer <mreimer@sdgsystems.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is a risk for memory leak in when something unexpected happens
and the function returns.
This was largely found by using a static code analysis program called cppcheck.
[fixed a typo of kfree() by tiwai]
Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eukrea-i.MX51 board was converted to use DT, ie we no longer have a
MACH_EUKREA_MBIMXSD51_BASEBOARD symbol.
Transformation of other boards planned for the near future, so this
patch removes all these dependencies and restricts build of this
driver to ARCH_MXC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch replaces the ssi specific functions write_ssi, read_ssi and
write_ssi_mask by standard regmap function calls.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder all variables in struct fsl_ssi_private to have groups that make
sense together. The patch also updates the struct documentation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The baudclock may be used and set by different streams.
Allow only the first stream to set the bitclock rate. Other streams have
to try to get to the correct rate without modifying the bitclock rate
using the SSI internal clock modifiers.
The variable baudclk_streams is introduced to keep track of the active
streams that are using the baudclock. This way we know if the baudclock
may be set and whether we may enable/disable the clock.
baudclock enable/disable is moved to hw_params()/hw_free(). This way we can
keep track of the baudclock in those two functions and avoid a running
clock while it is not used. As hw_params()/hw_free() may be called
multiple times for the same stream, we have to use baudclk_streams
variable to know whether we may enable/disable the clock.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In i2s master mode the fsl_ssi driver depends on someone calling
.set_tdm_slot correctly. In this mode though only a DC value of
2 is allowed, so set it in this case and no longer depend on
.set_tdm_slot.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The baudclock_locked variable is only used in functions which
are serialized anyway from the core. No need to have a lock
around the variable, so remove it.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The fsl_ssi driver uses the .set_sysclk callback to configure the
bitclock for master mode. This is unnecessary since the bitclock
is known in hw_params. This patch configures the bitclock from .hw_params.
.set_dai_sysclk now sets a bitclock frequency which is preferred over
the default calculated bitclock frequency.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The simple soundcard binding has its own way for specifying the dai
format. To be able to use this binding we have to make the fsl,mode
property optional. As the property is used in existing devicetrees
keep the option around for compatibility reasons.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Introduce a SoC data struct which contains the differences between
the different SoCs this driver supports. This makes it easy to support
more differences without having to introduce a new switch/case each
time.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use card PM ops from ASoC core instead of defining custom PM ops here since
we are calling anyway common suspend/resume callbacks.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify byt-rt5640.c and haswell.c machine drivers by using
devm_snd_soc_register_card(). Remove also needless dev_set_drvdata()
from byt_rt5640_probe() since snd_soc_register_card() does it too.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few files contain duplicate headers. This patch removes the second entry of
duplicate in each file under question.
There is no functional changes.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Stored DSP DMA pointer must be cleared before starting the stream since
PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is
updated. In that case last position of previous stream was wronly returned.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The new value argument needs proper shift to match the mask bit fields.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Baytrail and Haswell SST IPC don't stop the kernel thread in error and
cleanup path thus leaving orphan kernel thread behind in such a case.
Also while at it, fix one error path in sst-haswell-ipc.c that doesn't free
hsw->msg.
[Jarkko: I edited the commit log a little]
Signed-off-by: Imre Deak <imre.deak@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec.
Jack detect code from Kevin Strasser <kevin.strasser@intel.com>, GPIO
resolving from Mika Westerberg <mika.westerberg@linux.intel.com> and fixes
and cleanups from Liam Girdwood <liam.r.girdwood@linux.intel.com>.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some unknown reason the parameters for snd_soc_test_bits() were in wrong
order:
It was:
snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */
while it should be:
snd_soc_test_bits(codec, reg, mask, val);
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need
to give a proper dependency. For fixing and simplification, move the
selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section.
Then each driver doesn't have to select them but only
SND_FIREWIRE_LIB.
Reported-by: Jim Davis <jim.epost@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS A8JN with AD1986A codec seems following the normal EAPD in the
normal order (0 = off, 1 = on) unlike other machines with AD1986A.
Apply the workaround used for Toshiba laptop that showed the same
problem.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=75041
Cc: <stable@vger.kernel.org> [3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checks for CONFIG_SND_DEBUG_ROM were added in v2.5.5 but a Kconfig
symbol SND_DEBUG_ROM was never added. These checks have always
evaluated to false. Remove them and the printk()s they hide.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checks for CONFIG_BCM_CS4297A_CSWARM were added in v2.6.11. The related
Kconfig symbol was never added so these checks always evaluated to true.
Remove them.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
WM8804 can run with PLL frequencies of 256xfs and 128xfs for
most sample rates. At 192kHz only 128xfs is supported. The
existing driver selects 128xfs automatically for some lower
samples rates. By using an additional mclk_div divider, it
is now possible to control the behaviour. This allows using
256xfs PLL frequency on all sample rates up to 96kHz. It
should allow lower jitter and better signal quality. The
behavior has to be controlled by the sound card driver,
because some sample frequency share the same setting. e.g.
192kHz and 96kHz use 24.576MHz master clock. The only
difference is the MCLK divider.
Signed-off-by: Daniel Matuschek <daniel@matuschek.net>
Tested-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit c687200b9d.
Dropping the def association and sequence from pinconf comparing is a
bit risky, It will introduce a greater risk of catching unwanted
machines.
And in addition, so far no BIOS experts give us an explicit answer
whether it makes senses to compare these two fields or not.
For safety reason, we revert this commit.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There was a typo here so we return directly instead of freeing "hwinfo".
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We checked "err" earlier. These things seem to be left over code.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This issue was reported by coccicheck using the semantic patch
at scripts/coccinelle/api/resource_size.cocci
Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X81 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X61 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1381 and ADAU1781 audio
CODECs. The device is a low-power, 24-bit stereo audio CODEC with multiple
analog inputs and outputs, two digital microphone inputs and an I2S interface.
The device can be controlled either using I2C or SPI. The main difference
between the two variants is that the ADAU1781 has a freely programmable SigmaDSP
processor, while the ADAU1381 has a fixed function wind noise reduction filter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1361 and ADAU1761 CODECs.
The device is a a low-power, 24-bit stereo audio CODEC with multiple analog
input and outputs, one digital microphone input and an I2S interface. The device
can be controlled either via I2C or SPI. The main difference between the two
variants is that the ADAU1761 has a built-in SigmaDSP, while the ADAU1361 has
not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite
different in the analog domain. This patch adds support for the common parts of
the ADAU1X61 and ADAU1X81 CODECs.
The patch also restores some of the alphabetical order in the Makfile and
Kconfig.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently bebob driver apply Raw Audio Data channel (in IEC 61883-1:2002,
Multi Bit Linear Audio Data channel in IEC 61883-6:20005) to IEC 60958
Conformant Data channel because both fireworks and bebob based devices
can handle it by ignoring each label.
This patch improves a comment about this.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently mutex_unlock() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some variables were declared without static even if they're not referred
to from external files. This commit make them local symbols for better
information-hiding by file unit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds suffix to register values of each device, to supress 'sparse'
warnings. Additionally, this commit changes offset values with integer literal.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The second argument of snd_efw_command_get_sampling_rate() means sampling
rate and its type is 'unsigned int'. But 'int' variable is passed as parameter.
It's better to apply the same type for the variable as its argument.
Similally, the type of variable for snd_efw_command_get_clock_source() and
avc_bridgeco_get_plug_type() should be the same type as each argument.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally, I intent to this argument given with 'struct snd_pcm_runtime.rate'
or params_rate(). They return value of 'unsigned int'. So 'unsigned int' is
better for the type of this argument.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two modes for Fireworks, IEC 61883 compliant or Windows.
So it's better to use enum type instead of int to express the intension,
even if C language specification defines to handle enum variables as usual
integer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of memdup_user() should be passed to return correct error.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
__be32 value should not be assigned directly to bool value.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit fixes a big for loop count with array. The limitation of loop
count should be calcurated with the number of elements in the array, not
with the number of bytes.
Aditionally, this commit apply the same declaration as a prototype in header
for the array.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have defines for adsp messages best to consistently use them.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit e5d80e82e3 (ASoC: sgtl5000: Convert to use regmap directly) a
kernel oops is observed after a suspend/resume sequence.
The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no
longer a valid pointer.
Add the remaining register entries into sgtl5000_reg_defaults[] and remove
sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and
make the code simpler.
Tested on a im53-qsb board.
Reported-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When CONFIG_PM is disabled, the CONFIG_SND_HDA_POWER_SAVE_DEFAULT symbol
does not get defined, which causes a build error for the hda-tegra driver:
hda/hda_tegra.c:80:25: error: 'CONFIG_SND_HDA_POWER_SAVE_DEFAULT' undeclared here (not in a function)
static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
^
/git/arm-soc/sound/pci/hda/hda_tegra.c:235:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function]
static void hda_tegra_disable_clocks(struct hda_tegra *data)
^
This works around the problem by not referencing that macro
when CONFIG_PM is disabled. Instead, we assume that it's disabled
unconditionally and cannot be enabled at runtime.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Dylan Reid <dgreid@chromium.org>
Cc: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit 8c5178fca4 ("ALSA: Add params_width() helpers") introduces
a helper to get the sample width. Updating Samsung related sound
drivers to use this helper.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Having the binary ones complement operator in the new bitmak value makes the
code hard to read.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When headphones are plugged in, force AFG and node 0x02
("Headphone Playback Volume") to D0 to avoid pop noises.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add "depends on I2C" to shut up the build errors from randconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make including the omap-pcm.h outside sound/soc/omap more convenient.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow jack GPIO pins be defined also using GPIO descriptor-based interface
in addition to legacy GPIO numbers. This is done by adding two new fields to
struct snd_soc_jack_gpio: idx and gpiod_dev.
Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is
NULL and otherwise idx is the descriptor index within the GPIO consumer
device.
New function snd_soc_jack_add_gpiods() is added for typical cases where all
GPIO descriptor jack pins belong to same GPIO consumer device. For other
cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before
calling snd_soc_jack_add_gpios().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs
are still passed and requested using legacy GPIO numbers the driver
internals are converted to use GPIO descriptor API.
Motivation for this is to prepare soc-jack so that it will allow registering
jack GPIO pins using both GPIO descriptors and legacy GPIO numbers.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Failure to terminate this match table can lead to boot failures
depending on where the compiler places the match table.
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DMAC src/dst addr needs to be set from driver when DT case.
(It was set from SoC/DMAEngine code when non-DT case)
This patch adds rsnd_gen_dma_addr() to set DMAC src/dst addr.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting to use DMAEngine.
But, DMA slave channel name "tx", "rx" is not enough
in DT case.
Becuase, it has many ports and path combination.
This patch adds rsnd_dma_of_name() to find
DMA channel name, for example
memory to SSI0 is "mem_ssi0",
SSI0 to memory is "ssi0_mem",
SSI0 to SRC0 is "ssi0_src0",
SRC0 to SSI0 is "src0_ssi0",
SRC0 to DVC0 is "src0_dvc0"...
Renesas sound want to use PIO transfer mode for some reasons.
It will be PIO tranfer mode if device node doesn't have
DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver uses many modules (= SSI/SRC/DVC),
and each module had own name.
But, each module name can be used as several purpose,
like clock name, DMA name etc...
This patch uses common name for each module.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting Gen1/Gen2.
SRC probe can return error if it was unknown
generation.
Now, rsnd_src_non_ops is not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DT DMA support needs struct platform_device pointer,
and it can get struct device pointer from platform_device.
Save platform_device instead of device.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Driver needs to call of_node_put() after of_get_chile_by_name()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, gGuard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
This change fixes all Tegra machine drivers. By code inspection, I
believe some non-Tegra machine drivers have the same issue. I'll send a
patch for that separately, once this is reviewed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes sure a format string can never get processed into the worker
thread name from the device name.
Signed-off-by: Kees Cook <keescook@chromium.org>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some platforms require that the codecs mclk is a fixed multiplication
factor of the audio stream rate. Add a optional property to the
binding to hold this factor and implement a hw_params() function to
make use of it.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the clock divisor and multiplier NI, MI values for audio
sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful
for the Odroid X2/U2 boards when the codec works in master mode and its
MCLK clock is fed from the I2S CDCLK output.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
[s.nawrocki@samsung.com: edited the commit description]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In post commit, a quirk of this firmware about transactions is reported.
This commit apply a workaround for this quirk.
They often fail transactions due to gap_count mismatch. This state is changed
by generating bus reset.
The fw_schedule_bus_reset() is an exported symbol in firewire-core. But there
are no header for public. This commit moves its prototype from
drivers/firewire/core.h to include/linux/firewire.h.
This mismatch still affects bus management before generating this bus reset.
It still takes a time to call driver's probe() because transactions are still
often failed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just powering on, these devices below wait to download firmware.
- Firewire Audiophile
- Firewire 410
- Firewire 1814
- ProjectMix I/O
But firmware version 5058 or later, flash memory in the device stores the
firmware. So this driver can enable these devices by sending a certain cue to
load the firmware.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmwares for M-Audio Firewire 410/1814 and ProjectMix I/O has a quirk to
ignore MIDI messages in data blocks more than 8. This commit uses a flag which
Fireworks uses for a similar quirk.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
M-Audio Firewire 1814 has a quirk, ProjectMix I/O also has. They transmit
empty packet with wrong value of dbc incremented by 8 at high sampling rate.
According to IEC 61883-1, this value should be the same as the one in
previous packet.
This commit adds a flag named as CIP_EMPTY_HAS_WRONG_DBC. With flag, the value
of dbc in empty packet is overwittern by an expected value.
This is an example of this quirk:
CIP Header 0 CIP Header 1 Payload size
010D0000 9004F759 210
010D0010 90040B59 210
010D0020 90042359 210
01020028 9004FFFF 2 <-
010D0030 90043759 210
010D0040 90044B59 210
010D0050 90046359 210
01020058 9004FFFF 2 <-
010D0060 90047759 210
010D0070 90048B59 210
010D0080 9004A359 210
01020088 9004FFFF 2 <-
010D0090 9004B759 210
010D00A0 9004CB59 210
010D00B0 9004E359 210
010200B8 9004FFFF 2 <-
010D00C0 9004F759 210
010D00D0 90040B59 210
010D00E0 90042359 210
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support some models which M-Audio produces
with DM1000 but its firmware is special. They are:
- Firewire 1814
- ProjectMix I/O
They have heavily customized firmware. The usual operations can't be applied to
them. For this reason, this commit adds a model specific member to 'struct
snd_bebob' and some model specific functions. Some parameters are write-only so
this commit also adds control interface for applications to set them.
M-Audio special firmware quirks:
- Just after powering on, they wait to download firmware. This state is
changed when receiving cue. Then bus reset is generated and the device is
recognized as a different model with the uploaded firmware.
- They don't respond against BridgeCo AV/C extension commands. So drivers
can't get their stream formations and so on.
- They do not start to transmit packets only by establishing connection but
also by receiving SIGNAL FORMAT command.
- After booting up, they often fail to send response against driver's request
due to mismatch of gap_count.
This module don't support to upload firmware.
Tested-by: Darren Anderson <darrena092@gmail.com> (ProjectMix I/O)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support some models which M-Audio produces
with DM1000/DM1000E with usual firmware. They are:
- Firewire 410
- Firewire AudioPhile
- Firewire Solo
- Ozonic
- NRV10
- FirewireLightBridge
According to a person who worked in BridgeCo, some models are produced with
'Pre-BeBoB'. This means that these products were released before BeBoB was
officially produced, and later BeBoB specification was formed. So these models
have some quirks.
M-Audio usual firmware quirks:
- Just after powering on, 'Firewire 410' waits to download firmware. This
state is changed when receiving cue. Then bus reset is generated and the
device is recognized as a different model with the uploaded firmware.
- 'Firewire Audiophile' also waits to download firmware but its
vendor id/model id is the same as the one after loading firmware.
- The information of channel mapping for MIDI conformant data channel is
invalid against BridgeCo specification.
This commit adds some codes for these quirks but don't support to upload
firmware.
This commit also adds specific operations to get metering information. The
metering information also includes status of clock synchronization if the model
supports to switch source of clock.
The specification of FirewireLightBridge is unknown. So in this time, normal
operations are applied for this model.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Focusrite
produces with DM1000/BeBoB. They are:
- Saffire
- Saffire LE
- SaffirePro 10 I/O
- SaffirePro 26 I/O
This commit adds Focusrite specific operations:
1. Get source of clock
2. Get/Set sampling frequency
3. Get metering information
The driver uses these functionalities to read/write specific address by async
transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Yamaha produced
with DM1000/BeBoB. They are:
- GO44
- GO46
This commit adds Yamaha specific operations. To get source of clock, AV/C Audio
Subunit command is used.
I note that their appearances are similar to some models of TerraTec; 'Go44' is
similar to 'PHASE 24 FW' and 'GO46' is similar to 'PHASE X24 FW'. But their
combination of Audio/Music subunits is a bit different.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows this driver to support all of models which Terratec produced
with DM1000/BeBoB. They are:
- PHASE 24 FW
- PHASE X24 FW
- PHASE 88 Rack FW
- EWS MIC2
- EWS MIC4
- Aureon 7.1 Firewire
For Phase series, this commit adds a Terratec specific operation. To get source
of clock. AV/C Audio Subunit command is used.
For EWS series and Aureon, this module uses normal operations.
Tested-by: Maximilian Engelhardt <maxi@daemonizer.de> (PHASE 24 FW)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is for some devices which have its own operations or quirks.
Many functionality should be implemented in user land. Then this commit adds
functionality related to stream such as sampling frequency or clock source. For
help to debug, this commit adds the functionality to get metering information
if it's available.
To help these functionalities, this commit adds some AV/C commands defined in
'AV/C Audio Subunit Specification (1394TA).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback PCM samples.
When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback MIDI messages.
When no AMDTP streams are running, this module starts AMDTP stream at current
sampling rate for MIDI substream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds proc interface to get these information for debugging:
- firmware information
- stream formation
- current clock source and sampling rate
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Normal BeBoB firmware has a quirk. When receiving bus reset, it transmits
packets with discontinuous value in dbc field.
This causes two situation, one is to abort streaming by firewire-lib as a
result of detecting the discontinuity. Another is to call driver's .update()
because of bus reset. These two is generated independently. (The former
depends on isochronous stream and the latter depends on IEEE1394 bus driver.)
When BeBoB driver works with XRUN-recoverable applications, this situation
looks like stream_start_duplex() call followed by stream_update_duplex() call
because applications will call snd_pcm_prepare() immediately at XRUN.
To update connections and streams at first, this commit use completion. When
queueing error occurs, stream_start_duplex() is forced to wait maximum
1000msec. During this, when .update() is called, the completion is waken and
stream_start_duplex() is processed without breaking connections.
At bus reset, stream_start_duplex() shouldn't break/establish connections and
stream_update_duplex() should update connections because a caller of
fw_iso_resources_allocate() is responsible for calling
fw_iso_resources_update() on bus reset.
This commit also adds a flag, which has an effect to skip checking continuity
for first packet. This flag is useful for BeBoB quirk to start handling packets
during streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds management functionality for connections and streams.
BeBoB uses CMP to manage connections and uses AMDTP for streams.
This commit also adds some BridgeCo's AV/C extension commands. There are some
BridgeCo's AV/C extension commands but this commit just uses below commands
to get device's capability and status:
1.Extended Plug Info commands
- Plug Channel Position Specific Data
- Plug Type Specific Data
- Cluster(Section) Info Specific Data
- Plug Input Specific Data
2.Extended Stream Format Information commands
- Extended Stream Format Information Command - List Request
For Extended Plug Info commands for Cluster Info Specific Data, I pick up
'section' instead of 'cluster' from document to prevent from misunderstanding
because 'cluster' is also used in IEC 61883-6.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds two functionality for hwdep interface, adds two parameters for
this driver, add a node for proc interface.
To receive responses from devices, this driver already allocate own callback
into initial memory space in host controller. This means no one can allocate
its own callback to the address. So this driver must give a way for user
applications to receive responses.
This commit adds a functionality to receive responses via hwdep interface. The
application can receive responses to read from this interface. To achieve this,
this commit adds a buffer to queue responses. The default size of this buffer is
1024 bytes. This size can be changed to give preferrable size to
'resp_buf_size' parameter for this driver. The application should notice rest
of space in this buffer because this driver don't push responses when this
buffer has no space.
Additionaly, this commit adds a functionality to transmit commands via hwdep
interface. The application can transmit commands to write into this interface.
I note that the application can transmit one command at once, but can receive
as many responses as possible untill the user-buffer is full.
When using these interfaces, the application must keep maximum number of
sequence number in command within the number in firewire.h because this driver
uses this number to distinguish the response is against the command by the
application or this driver.
Usually responses against commands which the application transmits are pushed
into this buffer. But to enable 'resp_buf_debug' parameter for this driver, all
responses are pushed into the buffer. When using this mode, I reccomend to
expand the size of buffer.
Finally this commit adds a new node into proc interface to output status of the
buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This interface is designed for mixer/control application. To use hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback PCM samples.
When AMDTP stream is already running for PCM or the source of clock is not
internal, available sampling rate is limited at current one.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk to ignore MIDI messages in data blocks more than 8.
This commit adds a flag for this quirk and codes to skip 8 or more data
blocks to transfer MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to capture/playback MIDI messages.
When no AMDTP streams are running, this driver starts AMDTP stream for MIDI
stream at current sampling rate.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk to reset data block counter at bus reset.
This commit adds a flag of CIP_SKIP_DBC_ZERO_CHECK. This flag has an effect
to skip checking dbc continuity when dbc is zero.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks firmware version 5.5 reports fix interval for dbc in each packet.
For example, AudioFire4:
CIP0 CIP1 Payload
00070000 900484FF 72
00070008 9004A8FF 72
00070008 90FFFFFF 02
00070010 9004D0FF 72
00070018 9004C4FF 72
00070020 9004E8FF 72
00070020 90FFFFFF 02
00070028 900410FE 72
The interval of each dbc should be 16 except for empty packet but it's still 8.
This commit adds a flag for this quirk and codes to refer to a fixed value.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of Fireworks firmware, named as 'AudioFire9', seems to transmit
packets with wrong value of dbs. It's always 0x11 but actual size of
data block is different.
This commit adds a flag for this quirk and some codes to calculate
correct size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk for the value of dbc field in transmitted packets.
For Fireworks, dbc means the end of events in current packet. This is out
of specification.
For example, AudioFire4:
CIP0 CIP1 Payload
01070092 90FFFFFF 02
0107009A 9001E17B 3A <-
010700A2 9001F6E5 3A
010700A2 90FFFFFF 02
010700AA 9001104F 3A <-
010700B2 900125B9 3A
010700BA 90013B23 3A
010700BA 90FFFFFF 02
010700C2 9001548E 3A <-
010700CA 900169F8 3A
010700CA 90FFFFFF 02
010700D2 90018362 3A <-
010700DA 900198CC 3A
According to IEC 61883-1/6, a packet following to empty packet has the same
value for its dbc. But for Fireworks, it's incremented and empty packet has
the same value as previous packet in dbc field.
This commit adds a flag for Fireworks and some codes to checking dbc continuity.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks has a quirk to transmit empty packets with TAG0. This commit
adds handling this quirk for full duplex stream synchronization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks manages connections by CMP and can transmit/receive AMDTP streams
with a few quirks. This commit adds functionality to start/stop the streams.
Major Fireworks products don't support 'SYT-Match' clock source mode, except
for AudioFire12/8(till 2009 July) with firmware version 1.0. Already in
previous commit, this driver don't support such old firmwares. So this commit
adds support for non 'SYT-Match' clock source modes.
I note that this driver has a short gap for MIDI streams when starting PCM
stream. When AMDTP streams are running only for MIDI data and PCM data is
going to be joined at different sampling rate, then AMDTP streams are
stopped once and started again after changing sampling rate.
Unfortunately, Fireworks is not fully compliant to IEC 61883-1/6. Some commits
following to this commit add these quirks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireworks uses own command and response. This commit adds functionality to
transact and adds some commands required for sound card instance and kernel
streaming.
There are two ways to deliver substance of this transaction:
1.AV/C vendor dependent command for command/response
2.Async transaction to specific addresses for command/response
By way 1, I confirm AudioFire12 cannot correctly response to some commands with
firmware version 5.0 or later. This is also confirmed by FFADO. So this driver
implement way 2.
The address for response gives an issue. When this driver allocate own callback
function into the address, then no one can allocate its own callback function.
This situation is not good for applications in user-land. This issue is solved
in later commit.
I note there is a command to change the address for response if the device
supports. But this driver uses default value. So users should not execute this
command as long as hoping this driver works correctly.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for devices based on Fireworks. This driver
just creates/removes card instance according to callbacks.
Fireworks is a board module which Echo Audio produced. This module
consists of three chipsets:
- Communication chipset for IEEE1394 PHY/Link and IEC 61883-1/6
- DSP or/and FPGA for signal processing
- Flash Memory to store firmwares
Current supported devices:
- Mackie Onyx 400F/1200F
- Echo AudioFire12/8(until 2009 July)
- Echo AudioFire2/4/Pre8/8(since 2009 July)
- Echo Fireworks 8/HDMI
- Gibson Robot Interface pack/GoldTop
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds three commands, which may be used by some firewire device
drivers. These commands are defined in 'AV/C Digital Interface Command Set
General Specification Version 4.2 (2004006, 1394TA)'.
1. PLUG INFO command (clause 10.1)
2. INPUT PLUG SIGNAL FORMAT command (clause 10.10)
3. OUTPUT PLUG SIGNAL FORMAT command (clause 10.11)
By the command 1, the drivers can get the number of plugs for AV/C unit or
subunit.
By the command 2 and 3, the drivers can get/set sampling frequency.
The 'firewire-speakers' already uses INPUT PLUG SIGNAL FORMAT command to set
sampling rate. So this commit also affects the driver.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices based on BeBoB use this type of AV/C transaction.
'Deferred Transaction' is defined in 'AV/C Digital Interface Command Set
General Specification' and is used by targets to make a response deferred
during processing it.
If a target may not be able to complete a command within 100msec since
receiving the command, then the target shall return INTERIM response,
to which final response will follow later. CONTROL/NOTIFY commands are
allowed for deferred transaction.
In the specification, devices allow to send INTERIM response just one time.
But this commit allows to handle several INTERIM response with two reasons.
One reason is to simplify codes, and another reason is to prepare for
devices which is out of specification.
There is an issue. In the specification, the interval between INTERIM
response and final response is 'Unspecified interval'. The specification
depends on each subunit specification for this interval.
But we promise to finish this function for caller. In this reason, I use
FCP_TIMEOUT_MS for this interval. Currently it's 125msec. When we find
devices which needs more time for this interval, then let us add some codes
to apply more interval for 'Unspecified interval'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Plug Control Registers have two fields related to the number of established
connections, one is 'Broadcast connection counter' and another is
'Point-to-point connection counter'. The driver can know there are established
connections or not to check these fields.
This commit is for considering about JACK/FFADO streaming. Currently, when
JACK/FFADO starts its streaming to the device, cmp_connection_establish() is
failed expectedly. This seems to be enough but there are some devices which
needs to change sampling frequency before trying to establish connections.
For such devices, this functionality is needed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some macros, codes with condition of direction and new functions
to handle output connection. Once cmp_connection_init() is executed with its
direction, CMP input and output connection can be handled by the same way.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds 'direction' member to 'cmp_connection' structure to indicate
the direction of connection. This patch also adds 'direction' argument to
cmp_connection_init() function to determine the direction.
The cmp_connection_init() function is exported and used in snd-firewire-speakers
so this patch also affect it.
This patch just add them. Actual implementation will be done by followed
patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Referring to IEC 61883-1, oMPR and iMPR, oPCR and iPCR have some fields with
the same role in the same position. This patch renames some macros, variables
and function arguments with "i" in its prefix to reuse them between oMPR and
iMPR, oPCR and iPCR.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, drivers can bring XRUN state for PCM substreams when error to
queue packets or detecting discontinuity of packet. The application may try to
recover this state by calling snd_pcm_prepare().
Depending on each driver, .prepare() includes restart streaming. Then there
is a state that PCM substreams are running but isochronous contexts are
stopped. In this case, when .pointer() is called, it refers to error pointer.
This commit is for a prevention of this bug.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds common PCM constraints according to current firewire-lib
implementation.
1.Maximum width for each sample is limited by 24.
AM824 in IEC 61883-6 can deliver 24bit data.
2. Minimum time for period is 5msec.
Apply the old value. For shorter latency, further works are needed.
3. In blocking mode, frames per period/buffer is aligned to 32.
Each packet can include some frames depending on its sampling rate. In
blocking mode, the number equals to SYT_INTERVAL. Currently firewire-lib
can schedule snd_pcm_period_elapsed() for each packet. So, for accurate
PCM interrupt, the number of frames per period/buffer should be aligned
to SYT_INTERVAL.
Currently firewire-lib is lack of better rules to achieve this. So LCM of
each value of SYT_INTERVALs (=32) is applied. This can be improved for
further work.
[Fixed the compile error due to the missing "&" by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, AMDTP functionality in firewire-lib supports mapping
for PCM data channels. With this mapping, firewire-lib can obsolete
a flag, CIP_HI_DUALWIRE, but Dice driver still keeps dual wire mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices arrange the position of PCM/MIDI data channel in AMDTP packet.
This commit allows drivers to set channel mapping.
To be simple, the mapping table is an array with fixed length. Then the number
of channels for PCM is restricted by 64 channels.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Generally, the devices can synchronize to handle 'presentation timestamp'
in CIP packets. This commit adds functionality to pick up this timestamp from
in-packets transmitted by the device, then use it for out packets.
In current implementation, this module generated the timestamp by itself. This
is 'SYT Match' mode. Then drivers with this module acts as synchronization
master. This commit allows this module to act as synchronization slave.
This commit restricts this mechanism is only available in blocking mode because
handling the timestamp in non-blocking mode is more complicated than in
blocking mode.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For duplex streams with synchronization, drivers should pick up
'presentation timestamp' from in-packets and use the timestamp for
out-packets. This commit is preparation for this.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capturing/playbacking MIDI messages, this commit adds one MIDI conformant
data channel. This data channel has multiplexed 8 MIDI data streams. So this
data channel can transfer messages from/to 8 MIDI ports.
And this commit allows to set PCM format even if AMDTP streams already start.
I suppose the case that PCM substreams are going to be joined into AMDTP
streams when AMDTP streams are already started for MIDI substreams. Each
driver must count how many PCM/MIDI substreams use AMDTP streams to stop
AMDTP streams.
There are differences between specifications about MIDI conformant data.
About the multiplexing, IEC 61883-6:2002, itself, has no information. It
describes labels and bytes for MIDI messages and refers to MMA/AMEI RP-027
for 'successfull implementation'. MMA/AMEI RP-027 describes 8 MPX-MIDI data
streams for one MIDI conformant data channel. IEC 61883-6:2005 adds
'sequence multiplexing' and apply this way and describe incompatibility
between 2002 and 2005.
So this commit applies IEC 61883-6:2005. When we find some devices compliant
to IEC 61883-6:2002, then this difference should be handles as device quirk
in additional work.
About the number of bytes in an MIDI conformant data, IEC 61883-6:2002 describe
0,1,2,3 bytes. MMA/AMEI RP-027 describes 'MIDI1.0-1x-SPEED', 'MIDI1.0-2x-SPEED',
'MIDI1.0-3x-SPEED' modes and the maximum bytes for each mode corresponds to 1,
2, 3 bytes. The 'MIDI1.0-2x/3x-SPEED' modes are accompanied with 'negotiation
procedure' and 'encapsulation details' but there is no specifications for them.
So this commit implements 'MIDI1.0-1x-SPEED' mode for playback, but allows
to pick up 1-3 bytes for capturing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capturing PCM, this commit adds the functionality to handle in-stream.
This is also applied for dual-wire mode.
Currently, capturing 32bit samples are supported.
When the sequence of in-packet has discontinuity of dbc, in-stream isn't handled
and amdtp_streaming_error() returns true.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codes can be reused to handle in-stream. This commit adds new functions.
This commit also renames some functions to keep naming consistency.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds 'direction' member to amdtp_stream structure to indicate its
direction. This patch also adds 'direction' argument to amdtp_stream_init()
function to determine its direction.
The amdtp_stream_init() function is exported and used by firewire-speakers and
dice so this patch also affects them.
This patch just add them. Actual implementation will be done by followed
patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some macros instead of fixed value for AMDTP according to
IEC 61883-1/6. These macros will also be used by followed patches.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames some functions, a structure and its member to reuse them
in both AMDTP in/out stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just two members in the alc269_pin_fixup_tbl[] can cover more than
10 Dell laptop models.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot a machine have the same codec, but they have different default
pinconf setting just because the def association and sequence is
different, as a result they can't share a hda_pintbl[], to overcome
it, we don't compare def association and sequence in the pinconf
matching.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Normally, we match on pci ssid only. This works but needs new code
for every machine. To catch more machines in the same quirk, let's add
a new type of quirk, where we match on
1) PCI Subvendor ID (i e, not device, just vendor)
2) Codec ID
3) Pin configuration default
If all these three match, we could be reasonably certain that the
quirk should apply to the machine even though it might not be the
exact same device.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "fixup_forced" flag will indicate whether a specific fixup
(or nofixup) has been set by the user, to override the driver's
default.
This flag will help future patches.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If kmalloc() fails, warnings will be loud enough. We can safely just
return -ENOMEM in such cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent Intel H97/Z97 chipsets need the similar setups like other
Intel chipsets for snooping, etc. Especially without snooping, the
audio playback stutters or gets corrupted. This fix patch just adds
the corresponding PCI ID entry with the proper flags.
Reported-and-tested-by: Arthur Borsboom <arthurborsboom@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ensure i2s->op_clk is not used when clk_get() for this clock fails.
This prevents working with an incorrectly configured clock in some
conditions.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.
- Support for specifying aux CODECs in DT.
- Removal of the deprecated mux and enum macros.
- More moves towards full componentisation.
- Removal of some unused I/O code.
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers.
- Several drivers exposed directly in Kconfig for use with simple-card.
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350.
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Merge tag 'asoc-v3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.16
Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.
- Support for specifying aux CODECs in DT.
- Removal of the deprecated mux and enum macros.
- More moves towards full componentisation.
- Removal of some unused I/O code.
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers.
- Several drivers exposed directly in Kconfig for use with simple-card.
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350.
We should prefer `const struct pci_device_id` over
`DEFINE_PCI_DEVICE_TABLE` to meet kernel coding style guidelines.
This issue was reported by checkpatch.
A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@
- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;
// </smpl>
It has been tested by compilation.
Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' into asoc-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
clk_get in probe function can be safely replaced with devm_clk_get.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_platform with devm_snd_soc_register_platform
in samsung_asoc_dma_platform_register(). This makes the function
samsung_asoc_dma_platform_unregister() redundant. This is removed and
all its users are updated.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace snd_soc_register_card with devm_snd_soc_register_card.
With this change, we can delete the empty remove functions.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The analog-loopback causes the speaker noises even if it's set to zero
volume. As a simple workaround, just get rid of the loopback mixer.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=873704
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the AA-loopback path to get rid of the constant white noise
that can be heard when headphones are used.
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAPM routes for this board are complete, hence we can let the core take care
of disconnecting non-connected pins rather than doing it manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for setting source clock of the "Core CLK"
of the IIS Multi Audio Interface.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we are moving the mmp platform towards multiplatform support,
we have to stop including platform header files.
This changes the pxa-ssp sound driver file to no longer depend
on mach/hardware.h and mach/dma.h. The code using the definitions
from those headers is actually gone already, the only thing
that was still being used was the pxa_dma_desc typedef, which
we can easily work around by using the normal 'struct pxa_dma_desc'
name.
The pxa2xx-dma driver still uses this header, so we include it
explicitly there, which is ok because that is only used on pxa,
not on mmp.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let the ALC5623 codec be instantiated from DT. Add a simple binding
for the additional control register and the jack detect register.
Also, add a prompt to the Kconfig entry for this CODEC, so that it can
be selected. Since kirkwood-t5325.c will no longer be used, we need to
be able to enable the CODEC in the mvebu_v5_defconfig etc.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Acked-by: Jason Cooper <jason@lakedaemon.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsl_ssi_set_dai_sysclk will be called from fsl_ssi_hw_params in the
next patch. Move up to avoid forward declaration and to keep the next patch
more readable. No functional change.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the fsl-ssi unit is used in i2s slave mode, it is possible that the
SSI unit starts transmitting data on the wrong channel. This happens
because the SSI does not synchronize with the left-right-clock by
default.
This patch enables transmit enable synchronization.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are some variables defined in struct fsl_ssi_private that describe
states that are also described by other variables.
This patch adds some helper functions that return exactly the same
information based on available variables. This helps to clean up struct
fsl_ssi_private and remove them from the probe function.
It also removes some not really used variables (new_binding, name).
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder the probe function to be able to move the second imx-specific
block to the seperate imx probe function. The patch also removes some
comments/variables/code that are not used anymore or could be simply
replaced by other variables.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify dma DT property handling. fsl,ssi-dma-events is not used
anymore. It passes invalid data to imx_pcm_dma_params_init_data() which
copies some data into an imx dma struct. This struct is never used in
imx-dma or imx-sdma because of generic OF DMA handling. The
"fsl,ssi-dma-events" is not used anywhere in dts files.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move imx specific probe code to a seperate function. It reduces the
size of the probe() function and makes the code and error handling
easier to understand.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of creating a name using string manipulation functions, we can
simply use the device name for the DAI driver struct.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move all code that is only used for debugging to a seperate file. This
makes it easier to see what functions are used for debugging only.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The bits we have to clear when disabling are different when the other
stream is still active.
This patch fixes the calculation of new register values after disabling
one stream. It also adds a more detailed description of the new register
value calculation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.
Also fixes the following warnings from smatch:
sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.
While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
I missed a rename during the review process. Fix the
MODULE_DEVICE_TABLE to match the structure.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a driver for the HDA block in Tegra SoCs. The HDA bus is
used to communicate with the HDMI codec on Tegra124.
Most of the code is re-used from the Intel/PCI HDA driver. It brings
over only two of the module params, power_save and probe_mask.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases the pcm stream is closed while context has been
scheduled to be restored, causing a null pointer deref panic.
Cancel work to ensure stream does not get freed while work is
still active/pending.
Also, restoring the pcm context can be safely skipped after the
stream has been stopped. Check if pcm stream is still running
before restoring stream context to help pending work finish
more quickly in stream close path.
Signed-off-by: Kevin Strasser <kevin.strasser@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This looks like a copy-paste bug, fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core marks widgets as connected by default when they are
initialized in snd_soc_dapm_new_control() so there is no need to call
snd_soc_dapm_enable_pin() from machine driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end
snd_soc_instantiate_card() so there is no need to call it from machine
driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Just move field descriptions to the struct description in the kernel-doc
format. There is no functional change.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause()
unconditinally during device suspend. In case where DMA controller
doesn't support PAUSE/RESUME functionality, this call is not able
to stop the DMA controller. In this scenario, audio playback doesn't
resume after device resume.
Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes
the issue.
It has been tested with audio playback on Samsung platform having
PL330 DMA controller which doesn't support PAUSE/RESUME.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Delete unnecessary local variable whose value is always 0 and that hides
the fact that the result is always 0.
A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
local idexpression ret;
expression e;
position p;
@@
-ret = 0;
... when != ret = e
return
- ret
+ 0
;
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microphone bias level configuration register can configure voltage between
2.2 V and 2.8 V but doesn't manage is voltage on or off. Microphone bias
on/off state is controlled by "MICBIAS" DAPM widget.
Therefore there is no need to update bias voltage conditionally depending on
jack state each time when codec goes to SND_SOC_BIAS_ON state and setting
can be moved to max98090_probe() as driver currently doesn't support other
levels than 2.8 V.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
BCL is being configured in two places producing a warning message.
Make sure we only configure BCLK once and when we are master.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add ACPI ID for MAX98090 and ACPI 5 I2C device probing support.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure the cache is fully flushed at resume time.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we already have a memeber struct snd_sst_params.codec to fill this.
so removing duplicate instance
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the support to implement drain_notify in Intels mfld driver
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 0cac6fc3eb.
This comiit was dropped from rev2 and would not be required as it renames the
platform ops as well which is not required.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I managed to drop a change to stream ID setting from commit 49fee17816
("ASoC: Intel: Only export one Baytrail DAI") leading to non-working
simultaneous capture-playback since after one DAI conversion
rtd->cpu_dai->id + 1 will be the same for both playback and capture.
Use substream->stream + 1 like it was in original Liam's patch.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the %pap printk specifier to print resource_size_t variables. This
fixes warnings on platforms where resource_size_t has a different size
than int.
Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas@ideasonboard.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We don't need more than one DAI for Baytrail SST. Usage becomes also more
straightforward by grouping playback and capture streams under the same PCM
device.
[Jarkko: I made Liam's sst-baytrail-pcm.c change a few lines smaller and
squashed together with my byt-rt5640.c change]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Prepare for single Baytrail DAI playback/capture link by accessing PCM data
using stream ID instead of rtd->dev. Now rtd->dev is unique for playback
and capture since they are exported as separate DAIs but not once converted
to single DAI.
[Jarkko: Separated from another commit with updated commit log]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The return statement was not indented correctly. I lined up the
condition a bit as well.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The if condition here was supposed to return on error but the return
statement is missing. The effect is that the ->mixername is set to
"???" instead of "DT019X".
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
Adding missing pm ops so that audio playback works across
suspend and resume cycle.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Additionally to the ak4642 pll frequencies the ak4648 also supports 13MHz,
19.2MHz and 26MHz. This adds support for these frequencies.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently unused, this is done to let the driver distinguish between
the different supported codec types in later patches.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALC and ALC Zero crossing detection has been enabled unconditionally.
Add controls for this.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds DVC (Digital Volume Controller)
support which is member of CMD unit.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rsnd_mod_ops would like to come to use multi parameter.
modify macro to enable it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now, it can get rsnd_dai_stream pointer from rsnd_mod.
Remove duplicate parameter from rsnd_mod_ops
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SRC module needs ADINR register settings,
but, it has many similar xxx_ADINR register,
and needs same settings.
This patch adds rsnd_get_adinr() to sharing code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound supports only SRC/SSI,
but, other module will be supported.
This patch adds rsnd_path_parse() macro to share code
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The register CLASS_D_CONTROL_1 is marked as volatile because it contains
a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1
register. This causes problems for the "Speaker Switch" control, which
will report an error if the CODEC is suspended because it relies on a
volatile register.
To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and
manually keep the register cache in sync by updating both bits when
changing the mute status.
Reported-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Commit 10df350977 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is
enabled.") caused following regression in Baytrail SST:
baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed
baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware
Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with
the same dma_dev device what is now used in sst_fw_new() when allocating the
DMA buffer.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This follows the same idea than commit 10df350977
("ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.") by using only
ACPI device for all DMA allocations. Since DMA masking is already done in
firmware loading it can be removed from here.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA
postion buffer properly while DMA1 and DMA2 can work well. So this patch masks
the buggy DMA0 by keeping it as opened.
This is a tentative workaround, so keep the change small as Takashi suggested.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't need to change those dividers if bclk and mclk remains the same
directions and values.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to Reference Manual -- ESAI Initialization chapter, as the
standard procedure of ESAI personal reset, the PCRC and PRRC registers
should be remained in its reset value and then configured after T/RCCR
and T/RCR configurations's done but before TE/RE's enabling.
So this patch moves PCRC and PRRC settings to the end of hw_params().
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ESAI can only output EXTAL clock source directly. But for FSYS clock source,
ESAI can not output it without getting through PSR PM dividers.
So this patch adds an extra check in the code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The range here from 1 to 16 is confined to FP divider only while the
sck_div indicates if the calculation contains PSR and PM dividers. So
for the case using PSR and PM since the sck_div is true, the range of
ratio would simply become bigger than 16.
So this patch fixes the condition here and adds one line comments to
make the purpose here clear.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We may read an unexpected value when detemining which codec is attached.
In that case, either a unsupported codec is attached or something wrong
with I2C. The driver will not work properly on both cases. So we return
an error for that.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This was initially removed in commit 6423c1875 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad1 ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Suspend/resume support for the atmel-pcm-pdc driver was broken in commit
f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support"). It
essentially reverted the modifications done in commit 10cab262 ("ASoC: Change
how suspend and resume obtain the PCM runtime"). The suspend and resume handlers
at the beginning check if dai->runtime is not NULL, but dai->runtime is always
NULL, hence the code never runs. Considering that nobody noticed any problems in
the last 4 years since the code was broken and that the driver does not set
SNDRV_PCM_INFO_RESUME, which means applications are expected to stop and restart
the audio stream during suspend/resume, it is probably safe to assume that his
code is not needed and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when the DAPM context bias level is SUSPEND and the target bias level
is OFF dapm_pre_sequence_async() will first transition to PREPARE and
dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and
then to OFF.
This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE
when either going to ON or away from ON. This avoids the extra unnecessary
transitions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there is no support for compressed audio in Baytrail ADSP firmware
there is no need to leave it on during suspend since ALSA PCM buffers are
too small for leaving ADSP on for playing or recording.
Implement PM callbacks to Baytrail byt-rt5640.c machine driver that call
snd_soc_suspend and snd_soc_resume functions and unset the ignore_suspend
fields in DAI links.
This makes soc-core and ALSA core gracefully suspend and resume active
stream and call sst_byt_pcm_trigger() during suspend-resume cycle.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add suspend and resume support to Baytrail SST DSP. This is implemented by
unloading firmware modules and putting DSP into reset prior suspend and
restarting DSP again in normal boot state after resume.
Context restore for running streams is implemented by scheduling a work from
sst_byt_pcm_trigger() that will allocate a stream with existing parameters
and start it from last known buffer position before suspend.
[Jarkko: Squashed together 5 WIP patches from Liam and 1 from me]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Suspend/resume requires reloading FW to boot state so we need to also make
sure that the driver matches the FW state at boot.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have to save the physical address of extended firmware block in the
beginning of mailbox every time when we boot the DSP firmware since that
mailbox address is re-used after DSP firmware is running. Otherwise DSP
firmware will get bogus extended firmware block address during next DSP
boot.
Currently this is not problem but becomes when DSP runtime rebooting is
implemented. Prepare for that by moving extended firmware address saving
from sst_byt_init() to sst_byt_boot().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Stream start position will be needed in resume code. Prepare for it by
adding start offset argument to sst_byt_stream_start().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Baytrail ADSP stream IPC simplifies a little by moving IPC_IA_START_STREAM
construction and sending directly into sst_byt_stream_start() from
sst_byt_stream_operations(). This is because IPC_IA_START_STREAM is only
stream IPC with extra message data so this move saves a few code lines.
Main motivation for this is to prepare for passing stream start position
to sst_byt_stream_start() which will be needed in resume code.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is for preparing suspend/resume support but can give also more
safeguard against concurrent timestamp structure access between DSP firmware
and host.
Now DSP DMA pointer is sampled in each pcm pointer callback in
sst_byt_pcm_pointer() but that is unneeded since DSP updates the timestamp
period basis and can potentially be racy if sst_byt_pcm_pointer() is called
when DSP is updating the timestamp.
By taking DSP DMA pointer only after period elapsed IPC messages in
byt_notify_pointer() and returning stored hw pointer in
sst_byt_pcm_pointer() there is less risk for concurrent access.
The same stored hw pointer can be also used in suspend/resume code for
restarting the stream at the same position.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more
flexible") added generic support for on_val/off_val in the DAPM core. With this
in place there is no need anymore for having a special event callback for
SND_SOC_DAPM_REG() widgets.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If we find a widget who's stream name matches the name of a DAI widget then
thats the one it should be connected to. Based on the widget id we can say in
which direction the path should be. No need to go back to the DAI and check the
stream names.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We already know the widgets we want to connect, so use snd_soc_dapm_add_path()
instead of snd_soc_dapm_add_route() in snd_soc_dapm_new_pcm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We already know which two widgets should be connected, so use
snd_soc_dapm_add_path() instead of snd_soc_dapm_add_route() in
snd_soc_dapm_connect_dai_link_widgets().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit bd23c5b661.
The patch claims that the patch is necessary to avoid double prefix addition
when calling snd_soc_dapm_add_route() from snd_soc_dapm_connect_dai_link_widgets().
But snd_soc_dapm_add_route() is called with the card's DAPM context, which does
not have a prefix, which means there is no prefix that could be added a second
time.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm().
Also for CODEC to CODEC links the widgets are connected cross-over via a DAI
link widget, meaning that the capture widget of one CODEC will be connected to
the playback widget of the other and vice versa. Whereas
snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of
the CPU DAI to the playback widget of the CODEC DAI and the capture widget of
the CPU DAI to the capture widget of the CODEC DAI. So not skipping
CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create
incorrect connections between the two CODECs which will cause DAPM to detect
active paths where there are none and unnecessarily power up widgets.
Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.")
Cc: <stable@vger.kernel.org> (for 3.14+)
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be
SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit 4b68b4e1c5 (ASoC: Intel: split the pcm and compress to
different files) the compressed ops haven't been built causing link
failures on allyesconfig and making the driver unbuildable. Add the
object to the Makefile to fix that.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by Vinod Koul <vinod.koul@intel.com>
When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255,
SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292,
SID: 0x10280684), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
And on the machine with SID 0x10280684, and the Lineout and external
microphone should be routed to docking, this patch also fix this
problem.
BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this is for further updates to driver which supports DPCM :)
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
With DPCM we have media dai used and no seperate headset and speaker dai so
remove the speaker dai
The vibra is no longer supported thru audio, so remove
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For manging them and adding support for more platforms
Code move only
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
as this will be used in compressed split file in subsequent patch
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
to sst-mfld-platform-pcm.c so that we can split pcm and compress to different
files for upcoming changes to support more platforms
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add some SST API calls to unload and reload firmware modules. This can be used
by PM code to restore state and also allow modular FW to unload and release
memory blocks.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now, all platform is using new style rsnd_dai_platform_info.
Keeping compatibility is no longer needed.
We can cleanup code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
All platform which used old style was
switched to new style.
R-Car sound can remove old style clock support,
use device dependent clock now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Return statements in functions returning bool should use
true/false instead of 1/0.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch moves the private register settings from probe() to reg_default
struct.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This include file is about to disapear. In addition it is
useless for this code. So it is time to remove it.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Mark Brown <broonie@linaro.org>
The headphone and mic jacks on Thinkpad T440 are assigned to pins NID
0x16 and 0x19, respectively. These need to be set up manually by a
fixup.
Reported-and-tested-by: Joschi Brauchle <joschi.brauchle@tum.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d191bd8de8 ("ASoC: snd_soc_codec includes snd_soc_component") removed the
last user of the num_dai field. Also remove the field itself.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The global card list was removed in commit b19e6e7b7 ("ASoC: core: Use driver
core probe deferral"). The 'list' field of the snd_soc_card struct has been
unused since then. This patch removes the field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support") added
a per card list that keeps track of all the DAIs that have been registered with
the card, but the list has never been used. This patch removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Block offset calculations are done in the contiguous allocator so
are not required here.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Cirrus Logic Low Power Stereo I2C CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for RPDNEN, NSHHPEN, BRIDGOFF, CPWMEN and PNDLSL, and add DT
bindings to access them.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch fixes the following dereference check ordering.
sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746)
git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
git remote update asoc
git checkout 0b708c87f6
vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c
a4b12990 Mark Brown 2014-03-12 740 };
a4b12990 Mark Brown 2014-03-12 741
a4b12990 Mark Brown 2014-03-12 742 static int hsw_pcm_probe(struct snd_soc_platform *platform)
a4b12990 Mark Brown 2014-03-12 743 {
a4b12990 Mark Brown 2014-03-12 744 struct sst_pdata *pdata = dev_get_platdata(platform->dev);
a4b12990 Mark Brown 2014-03-12 745 struct hsw_priv_data *priv_data;
0b708c87 Liam Girdwood 2014-05-02 @746 struct device *dma_dev = pdata->dma_dev;
0b708c87 Liam Girdwood 2014-05-02 747 int i, ret = 0;
a4b12990 Mark Brown 2014-03-12 748
a4b12990 Mark Brown 2014-03-12 @749 if (!pdata)
a4b12990 Mark Brown 2014-03-12 750 return -ENODEV;
a4b12990 Mark Brown 2014-03-12 751
a4b12990 Mark Brown 2014-03-12 752 priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When using auto-muted controls it may happen that the register value will not
change when changing a control from enabled to disabled (since the control might
be physically disabled due to the auto-muting). We have to make sure to still
update the DAPM graph and disconnect the mixer input.
Fixes: commit 5729507 ("ASoC: dapm: Implement mixer input auto-disable")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The function is only used locally, make it static.
Fixes the following warning from sparse:
sound/soc/soc-core.c:1644:22: warning: symbol 'soc_find_matching_codec' was not declared. Should it be static?
Fixes: 3ca041ed ("ASoC: dt: Allow Aux Codecs to be specified using DT")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-By: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Other people would clearly understand each member and improve if they want.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
People would simply know what the driver gets the best for the current
sample rate playback.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The sysclk is one the clock sources that could be selected to derive
tx clock. But the route for sysclk is a bit different that it does
not only contain txclk df divider but also have an extra sysclk df.
So this patch mainly adds syclk df configuration support so as to
let the driver be able to get clock from sysclk.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We should have used _df by following the reference manual at the beginning.
So this patch just renames them.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The clock mux for the Freescale S/PDIF controller has eight clock sources
while most of them are from other moudles and even system clocks that do
not allow a rate-changing operation.
So we here only allow the clk_set_rate() and clk_round_rate() happened to
spdif root clock, the private clock for S/PDIF controller.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently infoframe contents and channel mapping are only set when a
sink (monitor) is present.
However, this does not make much sense, since
1) We can make a very reasonable guess on CA after 18e391862c ("ALSA:
hda - hdmi: Fallback to ALSA allocation when selecting CA") or by
relying on a previously valid ELD (or we may be using a
user-specified channel map).
2) Not setting infoframe contents and channel count simply means they
are left at a possibly incorrect state - playback is still allowed
to proceed (with missing or wrongly mapped channels).
Reasons for monitor_present being 0 include disconnected cable, video
driver issues, or codec not being spec-compliant. Note that in
actual disconnected-cable case it should not matter if these settings
are wrong as they will be re-set after jack detection, though.
Change the behavior to allow the infoframe contents and the channel
mapping to be set even without a sink/monitor, either based on the
previous valid ELD contents, if any, or based on sensible defaults
(standard channel layouts or provided custom map, sink type HDMI).
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: Stephan Raue <stephan@openelec.tv>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel
count") channel count is no longer being set if monitor_present is 0.
This is because setting the count was moved after the CA value is
determined, which is only after the monitor_present check in
hdmi_setup_audio_infoframe().
Unfortunately, in some cases, such as with a non-spec-compliant codec or
with a problematic video driver, monitor_present is always 0. As a
specific example, this seems to happen with gen1 ATV (SiI1390 codec),
causing left-channel-only stereo playback (multi-channel playback has
apparently never worked with this codec despite it reporting 8 channels,
reason unknown).
Simply setting converter channel count without setting the pin infoframe
and channel mapping as well does not theoretically make much sense as
this will just mean they are out-of-sync and multichannel playback will
have a wrong channel mapping.
However, adding back just setting the converter channel count even in
no-monitor case is the safest change which at least fixes the stereo
playback regression on SiI1390 codec. Do that.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Stephan Raue <stephan@openelec.tv>
Tested-by: Stephan Raue <stephan@openelec.tv>
Cc: <stable@vger.kernel.org> # 3.12+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the Realtek ALC5645 codec driver. It is the base
version that because the jack detect function is not implemented to
it, the headphone and AMIC1 are not workable. We will fill up the
further functions later.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data
for DSP modules
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Read the stream offset and presentation position from DSP memory rather
than using the old estimated position. This fixes timing issues with
pulseaudio.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
hw_params() can be called multiple times. Make sure we release the DSP
stream that was allocated on previous hw_params() calls before allocating
a new DSP stream.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Intel IOMMU requires that the ACPI device is used to allocate all
DMA memory buffers. This means we need to pass the DMA device pointer into child
component devices that allocate DMA memory.
We also only set the DMA mask for the ACPI device now instead of for each
component device.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix page table creation on Haswell and Broadwell to remove unsafe
virt_to_phys mappings and use more portable SG buffer. Use audio buffer
APIs to allocate DMA buffers.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we add the allocated blocks to the modules list of blocks.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we dont alloc blocks twice with requests spanning more
than one block.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that INPUT is required for the CS42L52 and WM8962 we can remove the
IS_ENABLED(INPUT) check in the drivers.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.
Add a workaround to detect and fix the corruption.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent addition of the USB audio mixer suspend/resume may lead to
deadlocks when the driver tries to call usb_autopm_get_interface()
recursively, since the function tries to sync with the finish of the
other calls. For avoiding it, introduce a flag indicating the resume
operation and avoids the recursive usb_autopm_get_interface() calls
during the resume.
Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The suspend callback of usb-audio driver may be called multiple times
per suspend when multiple USB interfaces are bound to a single sound
card instance. In such a case, it's superfluous to save the mixer
values multiple times. This patch fixes it by checking the counter.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This (widely used) construction:
if(printk_ratelimit())
dev_dbg()
Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.
[ 533.803964] retire_playback_urb: 852 callbacks suppressed
[ 538.807930] retire_playback_urb: 852 callbacks suppressed
[ 543.811897] retire_playback_urb: 852 callbacks suppressed
[ 548.815745] retire_playback_urb: 852 callbacks suppressed
[ 553.819826] retire_playback_urb: 852 callbacks suppressed
So use dev_dbg_ratelimited() instead of this construction.
Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 204dec93ea "ASoC: fsl: Allow to select individual common
options", it is possible to enable SND_SOC_FSL_SSI and SND_SOC_FSL_SPDIF
manually, either as loadable modules or built-in. This unfortunately
leads to a link error if one or both of them are built-in, while
the imx-pcm-dma framework is a loadable module:
sound/built-in.o: In function `fsl_ssi_probe':
:(.text+0x51fb8): undefined reference to `imx_pcm_dma_init'
sound/built-in.o: In function `fsl_spdif_probe':
:(.text+0x52e20): undefined reference to `imx_pcm_dma_init'
This changes Kconfig to prevent this case by using 'select' to turn
on the imx-pcm-dma code from both drivers. For consistency, we also
turn on the imx-pcm-fiq code, which is an alternative to the dma
implementation.
Note that imx-pcm-fiq is platform dependent, so we must not enable
that unless we are building a kernel for i.MX. Note also the
"if SND_IMX_SOC != n" syntax as opposed to the normal "if SND_IMX_SOC".
This is needed to avoid turning on the options as 'm' if 'SND_IMX_SOC'
is a module.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The cx20442 codec driver used here requires the TTY layer to
be enabled, or we get a link error:
sound/built-in.o: In function `cx20442_codec_remove':
cx20442.c:398: undefined reference to `tty_hangup'
sound/built-in.o: In function `ams_delta_remove':
ams-delta.c:613: undefined reference to `tty_unregister_ldisc'
sound/built-in.o: In function `ams_delta_cx20442_init':
ams-delta.c:559: undefined reference to `tty_register_ldisc'
This adds the missing dependency in the E3 configuration, there
was already one for the codec.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The WM8978 driver needs I2C to be enabled, so the
SND_SIU_MIGOR option also requires this.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This codec requires I2C to be enabled, so any other option
that selects it should also depend on I2C.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The WM8904 codec driver needs I2C to be enabled, so the
SND_ATMEL_SOC_WM8904 option also requires this.
Found using randconfig build testing.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will be useful for out-of-tree drivers since in-tree drivers
could select it automatically.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will be useful for out-of-tree drivers since in-tree drivers
could select it automatically.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The missing dependency can lead to build errors, so
make it explicit in Kconfig.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The UDA1380 driver needs I2C to be enabled, so
SND_SOC_SAMSUNG_H1940_UDA1380 and
SND_SOC_SAMSUNG_RX1950_UDA1380 also
require this.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The codec requires I2C to be enabled, so any other option
that selects it should also depend on I2C.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds device tree support to the Nokia N900 audio driver and
adds documentation for the DT binding.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for specifying auxiliary codecs and
codec configuration via device tree phandles.
This change adds new fields to snd_soc_aux_dev and snd_soc_codec_conf
and adds support for the changes to SoC core methods.
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add more error messages making it easier to identify problems.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Update the driver to get GPIO numbers from the
devm gpiod API instead of requesting hardcoded
GPIO numbers.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a preparation for DT based booting where the McBSP id
is set to -1 for all McBSP instances.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The symbol "nuc900_ac97_data" is used by the nuc900_pcm driver,
which may be a loadable module, so we should export it.
If one tries to build SND_SOC_NUC900 without SND_SOC_NUC900_AC97,
the kernel fails to link because of the reference to nuc900_ac97_data.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
dma_addr_t may be 64 bit wide, which causes a build failure
when doing a division on it. Here it is safe to cast to an
u32 type, which avoids the problem.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Tested-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This adds a missing dependency for SND_SOC_SMDK_WM8580_PCM to
require REGMAP_I2C to be enabled, avoiding possible build
erorrs.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This codec requires I2C to be enabled, so any other option
that selects it should also depend on I2C.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building ARM randconfig got into a situation where CONFIG_INPUT
is turned off and SND_SOC_ALL_CODECS is turned on, which failed
for two codecs trying to use the input subsystem. Some other
drivers also select one of these codecs and consequently need an
explicit dependency added.
Appending to the dependency list seems the easiest way out,
since this is not a practical limitation. If anyone really
needs to build these codecs for a kernel with no input support,
a more sophisticated solution can be implemented.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
49101a25ac "ASoC: cq93vc: Remove the set_cache_io() entirely from
ASoC probe" introduced the cq93vc_get_regmap function that has an
obvious build error referring to the 'codec' variable that is not
declared anywhere"
sound/soc/codecs/cq93vc.c: In function 'cq93vc_get_regmap':
sound/soc/codecs/cq93vc.c:157:34: error: 'codec' undeclared (first use in this function)
struct davinci_vc *davinci_vc = codec->dev->platform_data;
^
This changes the code to compile again, presumably in the way it was
intended. Not tested.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch correct rt5640's device ID
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x1028067e), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds devicetree support for mc13783-codec.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch converts the rx51 ASoC module to use
devm_snd_soc_register_card.
Signed-off-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add module alias to support driver autoloading.
Signed-off-by: Pali Rohár <pali.rohar@gmail.com>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Mark the array and the string const by using "static const char * const
foo[]" instead of "static const char* foo[]".
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Added machine driver to instantiate I2S based sound card on Snow
board. It has MAX98095 audio codec on board.
There are some other variants for Snow board which have MAX98090
audio codec. Hence support for MAX98090 is also added to this
driver.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The rxclk rate actually uses sysclk, ipg clock for example, as its
reference clock to calculate it. But the driver currently doesn't
pass a correct clock source. So fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Machine specific trigger callback allows to do final stream start/stop
related operations in a machine driver after setting up the codec, DMA and
DAI.
One example could be clock management for linked streams case where machine
driver can start/stop synchronously the linked streams.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Stefan Roese <sr@denx.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This fixes module unload regressions introduced by commits 98e639fb8a
("ASoC: Track which components have been registered with
snd_soc_register_component()") and b37f1d123c ("ASoC: Let snd_soc_platform
subclass snd_soc_component").
First commit causes component_list to be corrupted when removing codec and
second when removing platform. Reason for both is that components associated
with platform or codec are never removed from the list because for them
registered_as_component field in struct snd_soc_component is always false.
Now list becomes corrupted when snd_soc_unregister_platform() or
snd_soc_unregister_codec() frees the platform or codec structure and where
the associated struct snd_soc_component is embedded.
Fix these by moving component unregistration and cleanup to a new local
function __snd_soc_unregister_component() that takes component as its
argument.
Since component is known for platforms and codecs the
__snd_soc_unregister_component() can be called directly and
snd_soc_unregister_component() takes care to find and unregister only
components that were registered using snd_soc_register_component().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent commit (ca460f8652) changed the CORB RP reset procedure to
follow the specification with a couple of sanity checks.
Unfortunately, Nvidia controller chips seem not following this way,
and spew the warning messages like:
snd_hda_intel 0000:00:10.1: CORB reset timeout#1, CORBRP = 0
This patch adds the workaround for such chips. It just skips the new
reset procedure for the known broken chips.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In soc_dapm_stream_event() we have the same code twice, once for the codec_dai
and once for the cpu_dai. This patch factors the duplicated code out into a
separate function. This will make it easier to modify the implementation (since
there is only one place that needs to be updated) and also easier to add support
for more than two DAIs per DAI link.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The introduced functios check AC97 if it's ready for communication and
read data is valid.
Signed-off-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will help to maintain HW accessors and, for example, switch from the
direct I/O to MMIO which is more convenient for PCI devices.
Signed-off-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix format string mismatch in snd_seq_midisynth_register_port().
Argument type of p is unsigned int.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New codec ALC293/ALC3235 support multifunction jacks.
It used for menual select the input device.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x10280674), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the rt5639 support to the OF match table.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This was initially removed in commit 6479f15ad ("ASoC: Remove name_prefix unset
during DAI link init hack"), but was brought back in commit 503ae5e0 ("ASoC:
core: Add helpers for dai link and aux dev init") by accident. This patch
removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using devm_regulator_bulk_get() can make the code cleaner and smaller as we
do not need to call regulator_bulk_free() in the error and remove paths.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Restore correct parsing of dai-link subnodes with more explicit
implementation for applying the "simple-audio-card,"-prefix to
dai-link property and subnode names.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace rtd_get_codec_widget() and rtd_get_cpu_widget() by a simple
dai_get_widget() in preparation for DAI-multicodec support, per Lars
suggestion.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Separate DAI link and aux dev initialization in preparation for
DAI multicodec support.
Since aux dev will remain using single codecs but DAI links
will be able to support multiple codecs.
No functional change.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
[fparent@baylibre.com: Adapt to 3.14+]
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The current simple-card driver limits the DT node name to "sound".
Any of other names is forbidden while actually we should allow DT
to pass other node names.
And if this function is being called, the node must already have
the compatible "simple-audio-card" in DTB. So there should be no
need to check the name here.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now ASoC core is getting the intersection of supported formats not only
from CPU and CODEC dai's but also from DMA's. However, there should be
no specific width limitation from SDMA side.
So drop it. Otherwise, we would only support S16_LE format for all i.MX
platforms.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Regmap is able to enable/disable the core clock automatically each time
it's going to access the registers. But for DMA cases during playback or
recording, it's totally beyong control of regmap. So we have to open the
clock manually.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I suppose there is a possibility that hsw_notification_work() may run after
sst_hsw_stream_free() which can lead to a kernel crash since struct
sst_hsw_stream is freed at that point and
stream = container_of(work, struct sst_hsw_stream, notify_work) is not valid
when hsw_notification_work() is run.
Reported-by: Derek Basehore <dbasehore@chromium.org>
Reported-by: Wenkai Du <wenkai.du@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
audmux_debugfs_init() is marked as __init, but is called from imx_audmux_probe()
which is not marked as __init. This creates a section mismatch and a potential
runtime crash (if imx_audmux_probe() is called after the .init section was
dropped). This patch removes the __init annotation from audmux_debugfs_init(),
which fixes the following warning:
WARNING: sound/soc/built-in.o(.text+0x86960): Section mismatch in reference
from the function imx_audmux_probe() to the function
.init.text:audmux_debugfs_init()
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The commit e41975ed (ASoC: core: Fix the DAI name getting) added a break
within the "if (id < 0 || id >= pos->num_dai)" while the original design
of the search didn't break the loop if that condition contented but only
mark the ret error and let it go on to search the next component.
In a case like dmaengine which's not a dai but as a component sharing an
identical name with a dai, both the id and pos->num_dai here could be 0.
If we break the search, we may never find the dai we want as it might be
placed behind its dmaengine in the component list.
So this patch fixes the issue above by following the original design to
let the search carry on.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The AK4104 has only one power supply, called VDD. Enable it as long as
the codec is in use.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The properties like format, bitclock-master, frame-master,
bitclock-inversion, and frame-inversion should be common to the dais
connected with a dai-link. For bitclock-master and frame-master
properties to be unambiguous they need to indicate the mastering dai
node with a phandle.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Adds struct device_node **bitclkmaster and struct device_node **framemaster
function parameters. With the new syntax bitclock-master and frame-master
properties can explicitly indicate the dai-link bit-clock and frame masters
with a phandle. This patch also makes the minimal changes to simple-card
for it to work with the updated snd_soc_of_parse_daifmt(). Simple-card appears
to be the only user of snd_soc_of_parse_daifmt() for now.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
In exiting kernel, if DAIFMT flags are set in dai_link and I2S is
set to run in master mode, the I2S clocks are not getting configured
resulting in no output.
Existing code clears the current I2S clock settings during i2s_startup
and requires that the clocks are reconfigured. It then assumes that
sound-card driver would call snd_soc_dai_{set_sysclk/set_fmt} to
configure the root clock.
1. Since I2S clock settings remain fixed for a board, it would be better
to set the clocks once during sound-card probe.
2. Also if the DAIFMT flags are set in dai_link, snd_soc_dai_set_fmt is
called during DAI probe.
If both these conditions are true, then I2S clock remains unconfigured
during audio playback. Fix this by removing the code to clear
rclk_srcrate in i2s_startup. Instead, reset this during DAI probe.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow the jz4740 audio drivers to be build when CONFIG_COMPILE_TEST is selected.
This should improve the build test coverage. There is one small piece of
platform dependent code in the jz4740-i2s driver. It uses the DMA request type
constants which are defined in a platform specific header. We can solve this by
moving them from the platform specific header to the I2S driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The new GPIO descriptor API is now the preferred way for handling GPIOs. It also
allows us to separate the platform depended code from the platform independent
code (Which will make it possible to increase build test coverage of the
platform independent code).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
rsnd_soc_dai_trigger() will be called
after rsnd_dai_pointer_update() function
which is using rsnd_lock().
Thus, it should be called from outside of rsnd_lock().
Kernel will be hangup without this patch.
Special thanks to Kataoka-san
Reported-by: Ryo Kataoka <ryo.kataoka.wt@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is a race between sst_byt_stream_free() and sst_byt_get_stream()
if sst_byt_get_stream() called from sst_byt_irq_thread() context is
accessing the byt->stream_list while a stream is deleted from the list.
A stream is added to byt->stream_list in sst_byt_stream_new() and deleted in
sst_byt_stream_free(). sst_byt_get_stream() is always protected by
sst->spinlock, but the stream addition and deletion are not protected.
The patch adds spinlock to both stream addition and deletion.
[Jarkko: Same fix added to sst-haswell-ipc.c too]
Signed-off-by: Wenkai Du <wenkai.du@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Create an explicit of_match_table entry for MAX98095 codec. Also
add a binding Documentation for this compatible string.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_snd_soc_register_platform() is used in drivers which can be build as
modules, so it needs to be exported to avoid linkers errors like:
ERROR: "devm_snd_soc_register_platform" [sound/soc/omap/snd-soc-omap.ko] undefined!
ERROR: "devm_snd_soc_register_platform" [sound/soc/davinci/snd-soc-davinci.ko] undefined!
Fixes: 8931bf620 ("ASoC: Add resource managed snd_soc_register_platform()")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Makes the code a bit shorter and will also allow us to remove the drivers
remove() callback eventually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Rather than calling snd_soc_set_dai_fmt(), just set the dai_fmt field in the
dai_link struct. Both have the same effect, but the later is a bit shorter and
also allows us to remove the now unused init callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The routes for this sound card are fully specified, so set the fully_routed
flag. This allows us to remove the manual snd_soc_dapm_nc_pin() calls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver")
jz4740-pcm.c file, but neglected to remove the Makefile entries.
Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Reported-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There were occasional ADSP crash during reboot testing:
[ 11.883364] BUG: unable to handle kernel paging request at ffffc90121700000
[ 11.883380] IP: [<ffffffffc024d8bc>] sst_module_insert_fixed_block+0x24f/0x26d [snd_soc_sst_dsp]
[ 11.883397] PGD 7800b067 PUD 0
[ 11.883405] Oops: 0002 [#1] SMP
[ 11.886418] gsmi: Log Shutdown Reason 0x03
The virtual address, ffffc90121700000, was out of range. The virtual
address is calculated by adding LPE base address with an offset:
sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size);
The offset is calculated in sst_byt_parse_module, by subtraction of
two virtual addresses dsp->addr.fw_ext and dsp->addr.lpe:
block_data.offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe);
These virtual addresses are assigned by kernel from ioremap:
sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size);
sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size);
In current driver code, offset is defined as unsigned int32:
struct sst_module_data {
...
u32 offset; /* offset in FW file */
};
Most of the time kernel assigned virtual addresses with addr.fw_ext
greater than addr.lpe. But sometimes it was the other way round.
Fix the problem by declaring offset as signed int32_t.
Signed-off-by: Wenkai Du <wenkai.du@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
After moving the IO layer inside ASoC to the component level we can now easily
move the standard control helpers also to the component level. This allows to
reuse the same standard helper control implementations for other components.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC framework is in the process of migrating all IO operations to regmap.
regmap has its own more sophisticated tracing infrastructure for IO operations,
which means that the ASoC level IO tracing becomes redundant, hence this patch
removes them. There are still a handful of ASoC drivers left that do not use
regmap yet, but hopefully the removal of the ASoC IO tracing will be an
additional incentive to switch to regmap.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no unlocked version of soc_widget_update_bits_locked() and there is no
plan to introduce it in the near future, so drop the _locked suffix.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are no users of snd_soc_update_bits_locked() left and it is identical to
snd_soc_update_bits(). So it can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We currently have two very similar IO abstractions in ASoC, one for CODECs, the
other for platforms. Moving this to the component level will allow us to unify
those two. It will also enable us to move the standard kcontrol helpers as well
as DAPM support to the component level.
The new component level abstraction layer is primarily build around regmap.
There is a per component pointer for the regmap instance for the underlying
device. There are four new function snd_soc_component_read(),
snd_soc_component_write(), snd_soc_component_update_bits() and
snd_soc_component_update_bits_async(). They have the same signature as their
regmap counter-part and will internally forward the call one-to-one to regmap.
If the component it not using regmap it will fallback to using the custom IO
callbacks. This is done to be able to support drivers that haven't been
converted to regmap yet, but it is expected that this will eventually be removed
in the future once all component drivers have been converted to regmap.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have been using rt5640.c codec driver with RT5642 codec chip before commit
022d21f004 ("ASoC: rt5640: add rt5639 support"). That commits starts using
device ID reading in reset register for adding device specific controls and
routes runtime.
Now since device ID appears to be different between RT5640 and RT5642 the
driver doesn't add those controls and routes that are valid also on RT5642.
Fix this by adding a device ID found by debugging and minimal code for
supporting RT5642.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Remove the cleanup code related to the platform from the DAI drivers at the
same time to avoid breakage.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix the error added by commit:
ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading
Reported-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 9e1fda4ae158 ("ASoC: dapm: Implement mixer input auto-disable")
is trying to free the widget it allocated by snd_soc_dapm_new_control()
call in dapm_kcontrol_data_alloc() by adding kfree(data->widget) to
dapm_kcontrol_free().
This is causing a widget double free with auto-disabled DAPM kcontrols
in sound card unregistration because widgets are already freed before
dapm_kcontrol_free() is called.
Reason for that is all widgets are added into dapm->card->widgets list
in snd_soc_dapm_new_control() and freed in dapm_free_widgets() during
execution of snd_soc_dapm_free().
Now snd_soc_dapm_free() calls for different DAPM contexts happens before
snd_card_free() call from where the call chain to dapm_kcontrol_free()
begins:
soc_cleanup_card_resources()
soc_remove_dai_links()
soc_remove_link_dais()
snd_soc_dapm_free(&cpu_dai->dapm)
soc_remove_link_components()
soc_remove_platform()
snd_soc_dapm_free(&platform->dapm)
soc_remove_codec()
snd_soc_dapm_free(&codec->dapm)
snd_soc_dapm_free(&card->dapm)
snd_card_free()
snd_card_do_free()
snd_device_free_all()
snd_device_free()
snd_ctl_dev_free()
snd_ctl_remove()
snd_ctl_free_one()
dapm_kcontrol_free()
This wasn't making harm with ordinary DAPM kcontrols since data->widget is NULL for
them.
Fixes: 9e1fda4ae158 (ASoC: dapm: Implement mixer input auto-disable)
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
The CODEC's write callback can return a negative error code, make sure to pass
that on correctly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In preparation for componentisation of the kcontrol helpers use
snd_soc_kcontrol_codec() instead of snd_kcontrol_chip().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Version 4 of McASP is using omap-pcm as platform driver and the omap-pcm
platform need to be registered using the cpu dai's device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The omap-pcm no longer need to be a platform driver since all cpu_dai will
bind the platform to it's own device which we can use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same name/node for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that the platform driver is registered with the cpu_dai's device we
can use the same node for it instead of the hardwired name.
We can also remove the cpu_dai_name and platform_name from the dai_link
struct since we only support DT boot on OMAP4/5
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the same device for the platform driver when registering as the dai
driver. This will enable us to clean up some DT booted cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the same device for the platform driver when registering as the dai
driver. This will enable us to clean up some DT booted cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Assign the dai dma data at dai driver probe time, not in startup.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
With the new calls it is going to be possible to bind the platform driver
to a dai device which makes it easier for us in a long run to handle DT
boots, and opens the possibility to move machine driver to generic simple
card.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the same device for the platform driver when registering as the dai
driver. This will enable us to clean up some DT booted cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Assign the dai dma data at dai driver probe time, not in startup.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the same device for the platform driver when registering as the dai
driver. This will enable us to clean up some DT booted cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Assign the dai dma data at dai driver probe time, not in startup.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify error handling and remove repetitive (and rarely executed) code
for unregistration by providing a devm_snd_soc_register_platform()
platform.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
PTR_RET is deprecated. PTR_ERR_OR_ZERO should be used instead. However,
we already know that IS_ERR is true, and thus PTR_ERR_OR_ZERO would
never yield zero, so we can use PTR_ERR here.
Signed-off-by: Christoph Jaeger <christophjaeger@linux.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix an incorrect sizeof() usage in sst_hsw_stream_get_volume(). sst_dsp_read()
is called to read into a variable of type u32, but is passed sizeof(u32 *) for
argument 'size_t bytes'. Detected by Coverity: CID 1195260.
Signed-off-by: Christian Engelmayer <cengelma@gmx.at>
Signed-off-by: Mark Brown <broonie@linaro.org>
The intent was to say "sizeof(*pos)" and not "sizeof(pos)".
The sizeof(*pos) is 8 bytes so the bug won't show up on 64 bit systems.
The sizeof(*dx) is 172 bytes so that will be a bugfix.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The REVID mask was changed by commit a1253ef6d3
"ASoC: cs42l51: split i2c from codec driver". Fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch change reg's type from unsigned short to int.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current code missed a gpio_free() call in cs42l73_i2c_remove().
Convert to use devm_gpio_request_one() to fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current code missed a gpio_free() call in cs42l52_i2c_remove().
Convert to use devm_gpio_request_one() to fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It should use STC_SYSCLK_DIV_OFFSET. Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some reason, some of the comments were actually in poorly encoded french.
Translate them in english like they should have been in the first place.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code contained in these sections are only dev_dbg calls, that are already
removed whenever DEBUG isn't defined.
Remove the redundant constructs.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some code was never compiled because hidden between an #if 0 .. #endif
structure, and even when removing these, it was never actually used elsewhere.
Remove it entirely.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit be4e6d3c0f ("ALSA: lx6464es: Use standard printk helpers") converted
the custom printk helpers that were used before to standard dev_* functions.
One of the dev_dbg calls had a typo, that was hidden away by an #if 0 .. #endif
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move to using the BIT macro for a few defines. It also allows to discard the
french comment that was saying exactly what the BIT macro is now pointing out.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit f9367f3fbe ("ALSA: lx6464es: Remove unused
function in pci/lx6464es/lx_core.c") removed the
lx_dsp_es_check_pipeline function that was the only user
of these defines.
Since they're useless now, simply remove them.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes the i2c bus code from the codec driver and creates seperate i2c driver.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
BugLink: http://bugs.launchpad.net/bugs/1305480
The kerneloops-daemon scans dmesg for common crash signatures, among
which is 'BUG:'. The message emitted by the PCM library is really a
warning, so the most expedient thing to do seems to be to change the
string.
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x1028067f), the headset mic can't be detected, after apply this
patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 0cd257bf9b, "ASoC: alc5623:
Convert to direct regmap API usage" broke probing of the codec,
because of wrong endinness of the ID and codec version read from the
device. Fix this by removing the existing flipping of the endiannes,
and extracting the codec type byte from the word from the regmap.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the second tlv320aic3x instance fails to
be probed from DT if the reset pin is shared with
the first one.
This patch fixes it by moving the list add of the
reset pin into the i2c_probe method.
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Remove a self assignment in sst_mem_block_alloc_scratch(). When calculating
buffer sizes there is no need for statements without effect. Detected by
Coverity: CID 1195249.
Signed-off-by: Christian Engelmayer <cengelma@gmx.at>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fix an incorrect sizeof() usage in sst_hsw_stream_get_volume(). sst_dsp_read()
is called to read into a variable of type u32, but is passed sizeof(u32 *) for
argument 'size_t bytes'. Detected by Coverity: CID 1195260.
Signed-off-by: Christian Engelmayer <cengelma@gmx.at>
Signed-off-by: Mark Brown <broonie@linaro.org>
Hopefully fixing a build failure reported by Stephen Rothwell - though
quite why the other OF headers don't include this as well I'm not sure.
Signed-off-by: Mark Brown <broonie@linaro.org>
The macro's name is SOC_ENUM, not SOC_VALUE.
Fixes: e13dd8ce ("ASoC: wm8997: Replace usage deprecated MUX/ENUM macros")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
By doing this, the driver can drop around 50 lines and become neater.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the precise definition of 'ret', which will be used for
the error check.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Patch ASoC: fsl_sai: Fix buggy configurations in trigger() doesn't entirely
fix the condition: FRDE of the current substream direction is being cleared
while the code is still using the non-updated one.
Thus this patch fixes this issue by checking the opposite one's FRDE alone
since the current one's is absolutely disabled.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This register is not defined in TI81xx and on AM335x/AM437x it is the
SYSCONFIG register which should not be touched by drivers since it is
related to PM and handled by the generic PM code.
This register write was there since the first time the davinci-mcasp driver
was appeared in the kernel.
The reason why it did not caused any issues on AM335x/AM437x is that it sets
bit 1 in SYSCONFIG register which in turn will enable the smart-idle mode.
This is the default mode and this is the mode McASP should be in also when
in use.
On TI81xx the register is not defined.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_VALUE_ENUM, SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated
and merely an alias for SOC_ENUM, SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace
the deprecated macros so we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VIRT_MUX and SOC_DAPM_ENUM_VIRT are deprecated and merely an alias
for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so we can
eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VIRT_MUX, SND_SOC_DAPM_VIRT_MUX_E and SOC_DAPM_ENUM_VIRT are
deprecated and merely an alias for SND_SOC_DAPM_MUX, SND_SOC_DAPM_MUX_E and
SOC_DAPM_ENUM. Replace the deprecated macros so we can eventually remove their
definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_VALUE_ENUM is deprecated and merely an alias for SOC_EMUM. Replace the
deprecated macros so we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_VALUE_ENUM, SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated
and merely an alias for SOC_ENUM, SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace
the deprecated macros so we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_VALUE_ENUM is deprecated and merely an alias for SOC_ENUM. Replace the
deprecated macro so we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VIRT_MUX and SOC_DAPM_ENUM_VIRT are deprecated and merely an alias
for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so we can
eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VIRT_MUX and SOC_DAPM_ENUM_VIRT are deprecated and merely an alias
for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so we can
eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_VALUE_MUX and SOC_DAPM_VALUE_ENUM are deprecated and merely an
alias for SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the deprecated macros so
we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_VALUE_ENUM, SND_SOC_DAPM_VIRT_MUX and SOC_DAPM_ENUM_VIRT are deprecated and
merely an alias for SOC_ENUM, SND_SOC_DAPM_MUX and SOC_DAPM_ENUM. Replace the
deprecated macros so we can eventually remove their definition.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add codec registration specific function in preparation
for DAI-multicodec support.
No functional change.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
[fparent@baylibre.com: Adapt to 3.14+]
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a helper for DAI widgets linking in preparation for
DAI-multicodec support.
No functional change.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
[fparent@baylibre.com: Adapt to 3.14+]
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add helper functions for codec DAI probe and remove in
preparation for DAI-multicodec support.
No functional change.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
[fparent@baylibre.com: Adapt to 3.14+]
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add dedicated helpers for codec and codec_dai search in preparation
for DAI-multicodec. It will help reducing the extra indentation
that will be introduced by the iteration over multiple codecs.
Previous implementation unnecessarily kept searching for a matching
codec in the remaining register codecs even if it was already found.
Fix that by returning in case of matching.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
[fparent@baylibre.com: Adapt to 3.14+]
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_match_ptr is defined in of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_match_ptr is defined in of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_match_ptr is defined in of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_match_ptr is defined in of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_match_ptr is defined in of.h. Include it explicitly.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes the following compilation warnings:
sound/soc/codecs/pcm512x.c:520:12: warning: ‘pcm512x_suspend’ defined but not used [-Wunused-function]
sound/soc/codecs/pcm512x.c:545:12: warning: ‘pcm512x_resume’ defined but not used [-Wunused-function]
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TI STA350 is an integrated 2.1-channel power amplifier that is
controllable over I2C. This patch adds an ASoC driver for it.
At a glance, this chip is very similar to the STA320 for which a driver
already exists. In details, however, the register maps contain subtle
differences which made a whole new driver easier to write and maintain.
[daniel@zonque.org: cleanups, DT property rework, rebased on asoc-next]
Signed-off-by: Sven Brandau <brandau@gmx.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The port driver only used to register component and dmaengine pcm.
Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the string "rt5639" to the list of I2C device IDs.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch moves the idle_bias_off setting to struct "soc_codec_dev_rt5640".
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch removes the unused or incorrect setting of clock source.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to identify clearly, the patch renames the function
"check_sysclk1_source" to "is_sys_clk_from_pll".
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In order to prevent the redundant memory usage, the pre-allocated size of
reg_default should be remove.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Nicolin Chen