When McASP is bit clock and frame clock master enable pin output for rx clocks.
Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update dapm_clock_event to use clk_prepare_enable and
clk_disable_unprepare.
Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pull media update from Mauro Carvalho Chehab:
- OF documentation and patches at core and drivers, to be used by for
embedded media systems
- some I2C drivers used on go7007 were rewritten/promoted from staging:
sony-btf-mpx, tw2804, tw9903, tw9906, wis-ov7640, wis-uda1342
- add fimc-is driver (Exynos)
- add a new radio driver: radio-si476x
- add a two new tuners: r820t and tuner_it913x
- split camera code on em28xx driver and add more models
- the cypress firmware load is used outside dvb usb drivers. So, move
it to a common directory to make easier to re-use it
- siano media driver updated to work with sms2270 devices
- several work done in order to promote go7007 and solo6x1x out of
staging (still, there are some pending issues)
- several API compliance fixes at v4l2 drivers that don't behave as
expected
- as usual, lots of driver fixes, improvements, cleanups and new device
addition at the existing drivers.
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (831 commits)
[media] cx88: make core less verbose
[media] em28xx: fix oops at em28xx_dvb_bus_ctrl()
[media] s5c73m3: fix indentation of the help section in Kconfig
[media] cx25821-alsa: get rid of a __must_check warning
[media] cx25821-video: declare cx25821_vidioc_s_std as static
[media] cx25821-video: remove maxw from cx25821_vidioc_try_fmt_vid_cap
[media] r820t: Remove a warning for an unused value
[media] dib0090: Fix a warning at dib0090_set_EFUSE
[media] dib8000: fix a warning
[media] dib8000: Fix sub-channel range
[media] dib8000: store dtv_property_cache in a temp var
[media] dib8000: warning fix: declare internal functions as static
[media] r820t: quiet gcc warning on n_ring
[media] r820t: memory leak in release()
[media] r820t: precendence bug in r820t_xtal_check()
[media] videodev2.h: Remove the unused old V4L1 buffer types
[media] anysee: Grammar s/report the/report to/
[media] anysee: Initialize ret = 0 in anysee_frontend_attach()
[media] media: videobuf2: fix the length check for mmap
[media] em28xx: save isoc endpoint number for DVB only if endpoint has alt settings with xMaxPacketSize != 0
...
* patchwork: (831 commits)
[media] cx88: make core less verbose
[media] em28xx: fix oops at em28xx_dvb_bus_ctrl()
[media] s5c73m3: fix indentation of the help section in Kconfig
[media] cx25821-alsa: get rid of a __must_check warning
[media] cx25821-video: declare cx25821_vidioc_s_std as static
[media] cx25821-video: remove maxw from cx25821_vidioc_try_fmt_vid_cap
[media] r820t: Remove a warning for an unused value
[media] dib0090: Fix a warning at dib0090_set_EFUSE
[media] dib8000: fix a warning
[media] dib8000: Fix sub-channel range
[media] dib8000: store dtv_property_cache in a temp var
[media] dib8000: warning fix: declare internal functions as static
[media] r820t: quiet gcc warning on n_ring
[media] r820t: memory leak in release()
[media] r820t: precendence bug in r820t_xtal_check()
[media] videodev2.h: Remove the unused old V4L1 buffer types
[media] anysee: Grammar s/report the/report to/
[media] anysee: Initialize ret = 0 in anysee_frontend_attach()
[media] media: videobuf2: fix the length check for mmap
[media] em28xx: save isoc endpoint number for DVB only if endpoint has alt settings with xMaxPacketSize != 0
...
Conflicts:
drivers/media/pci/cx25821/cx25821-video.c
drivers/media/platform/Kconfig
Current code does this:
be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1])
Which is effectively (neglecting the index):
be16_to_cpu(be16_to_cpu(*((u16 *) buf)))
This means the int16 in the buffer is not converted at all.
Daniel Mack confirmed that the driver works on little endian
CPUs, leading to the conclusion that the device-side structure
is actually little endian.
This changes the code to use le16_to_cpu().
Caught by sparse.
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here's the big char / misc driver update for 3.10-rc1
A number of various driver updates, the majority being new functionality
in the MEI driver subsystem (it's now a subsystem, it started out just a
single driver), extcon updates, memory updates, hyper-v updates, and a
bunch of other small stuff that doesn't fit in any other tree.
All of these have been in linux-next for a while
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'char-misc-3.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc
Pull char/misc driver update from Greg Kroah-Hartman:
"Here's the big char / misc driver update for 3.10-rc1
A number of various driver updates, the majority being new
functionality in the MEI driver subsystem (it's now a subsystem, it
started out just a single driver), extcon updates, memory updates,
hyper-v updates, and a bunch of other small stuff that doesn't fit in
any other tree.
All of these have been in linux-next for a while"
* tag 'char-misc-3.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc: (148 commits)
Tools: hv: Fix a checkpatch warning
tools: hv: skip iso9660 mounts in hv_vss_daemon
tools: hv: use FIFREEZE/FITHAW in hv_vss_daemon
tools: hv: use getmntent in hv_vss_daemon
Tools: hv: Fix a checkpatch warning
tools: hv: fix checks for origin of netlink message in hv_vss_daemon
Tools: hv: fix warnings in hv_vss_daemon
misc: mark spear13xx-pcie-gadget as broken
mei: fix krealloc() misuse in in mei_cl_irq_read_msg()
mei: reduce flow control only for completed messages
mei: reseting -> resetting
mei: fix reading large reposnes
mei: revamp mei_irq_read_client_message function
mei: revamp mei_amthif_irq_read_message
mei: revamp hbm state machine
Revert "drivers/scsi: use module_pcmcia_driver() in pcmcia drivers"
Revert "scsi: pcmcia: nsp_cs: remove module init/exit function prototypes"
scsi: pcmcia: nsp_cs: remove module init/exit function prototypes
mei: wd: fix line over 80 characters
misc: tsl2550: Use dev_pm_ops
...
Some Asihpi formats are not supported or invalid, and their mapping to
ALSA format is set to -1.
Before performing the format conversion into ALSA bitwise formats,
add a consistency check for the requested format, as done in
snd_card_asihpi_playback_formats().
Compile tested only.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
for reading compressed data, we need to allow when we are paused, draining or
stopped.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Cc: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Cc: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This enables better volume controls than the current model parser.
Also, because the original quirk for X220 was added to fix
docking station support, add the TP410 fixup instead.
Reported-by: Willian Jon McCann <william.jon.mccann@gmail.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes in the USB API ("implement new semantics for
URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the
default, and changed this flag to mean that URBs can be delayed.
This is not the behaviour wanted by any of the audio drivers because
it leads to discontinuous playback with very small period sizes.
Therefore, our URBs need to be submitted without this flag.
Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org>
Cc: <stable@vger.kernel.org> # 3.8 only
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merging in fixes since there's a conflict in the omap4 clock tables caused by
it.
* fixes: (245 commits)
ARM: highbank: fix cache flush ordering for cpu hotplug
ARM: OMAP4: hwmod data: make 'ocp2scp_usb_phy_phy_48m" as the main clock
arm: mvebu: Fix the irq map function in SMP mode
Fix GE0/GE1 init on ix2-200 as GE0 has no PHY
ARM: S3C24XX: Fix interrupt pending register offset of the EINT controller
ARM: S3C24XX: Correct NR_IRQS definition for s3c2440
ARM i.MX6: Fix ldb_di clock selection
ARM: imx: provide twd clock lookup from device tree
ARM: imx35 Bugfix admux clock
ARM: clk-imx35: Bugfix iomux clock
+ Linux 3.9-rc6
Signed-off-by: Olof Johansson <olof@lixom.net>
Conflicts:
arch/arm/mach-omap2/cclock44xx_data.c
The Scarlett 2i2 seems to take almost 500 ms to set the sample rate,
even if the clock is currently set to that value. This patch speeds
up prepare of the device, by avoiding setting the clock to something
it already is.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The virt_to_bus/bus_to_virt functions have been deprecated
for as long as I can remember, and they are used in very
few remaining instances, usually in obscure ISA device
drivers. The OSS sound drivers are the only ones that are
still used on the ARM architecture, and only on some of
the earliest StrongARM machines.
The problem for converting the OSS subsystem to use
dma_map_single instead is that the caller of virt_to_bus
does not have a device pointer, since the subsystem has
never been ported to use the common device infrastructure.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ARM cannot handle udelay for more than 2 miliseconds, so we
should use mdelay instead for those.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes, nothing too major though the DMA changes fix modular
builds.
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Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
A few more fixes, nothing too major though the DMA changes fix modular
builds.
We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:
ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
....
It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card->probing = 1.
Since the call itself during card->probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.
Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.
There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.
When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.
Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Torstein Hegge <hegge@resisty.net>
Reported-and-tested-by: Yves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We get a lot of build warnings from the msp driver like:
In file included from sound/soc/ux500/ux500_msp_dai.h:21:0,
from sound/soc/ux500/mop500.c:25:
sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: 'struct msp_i2s_platform_data' declared inside parameter list [enabled by default]
struct msp_i2s_platform_data *platform_data);
^
sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default]
The easiest solution is to add a declaration of the struct name.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length).
I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels.
Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback"
number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dma_request_slave_channel() is a more appropriate API for dmaengine
clients that adopt generic DMA bindings to call. Let's use it instead
of of_dma_request_slave_channel() to save <linux/of_dma.h> include.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The examples in Documentation/devicetree/bindings/dma/dma.txt recommends
the name for dma channel doing both RX and TX to be "rx-tx". This
becomes a common pattern that has been adopted by platforms that
converts to generic DMA bindings. Let's follow this common pattern in
generic-dmaengine-pcm.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These are being reported as being so noisy at high mic boost levels,
so they are unusable in practice.
Therefore artificially limit the boosts.
BugLink: https://bugs.launchpad.net/bugs/1089795
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting mxs to generic DMA engine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MXS SAIF is only half-duplex so set the SNDRV_PCM_INFO_HALF_DUPLEX flag for
the PCM in order to prevent playback and capture from running at the same time.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some platforms which are half-duplex share the same DMA channel between the
playback and capture stream. Add support for this to the generic dmaengine PCM
driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set the timeout for USB control set messages according to the USB 2
spec, using the macros from include/linux/usb.h.
The get timout becomes 5000 ms even though it is 500 ms in the
spec. This patch is required to run the Hercules RMX2 which needs a
timeout of 1240 ms.
More notes from author:
I still distinguish between set and get but as long both are 5000 ms
GCC will remove it anyway. IMHO this is more easy read and there is no
need to explain why we use a get timeout for set messages.
Signed-off-by: Daniel Schürmann <daschuer@mixxx.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous patch by Dan, we should clear the data to be
returned from certain compress ioctls, namely,
snd_compr_get_codec_caps() and snd_compr_get_params().
This time, we can simply replace kmalloc() with kzalloc().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the ->get_caps() function doesn't clear the buffer then there would
stack information leaked to userspace. For example,
soc_compr_get_caps() can return success without clearing the buffer.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously we just hard coded all streams as playback streams, this
patch checks the DAI to see if it is a capture or playback stream. It is
worth noting that at this time only unidirectional streams are
supported.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The app_pointer is managed locally by the compress core for memory
mapped DSPs but for DSPs that are not memory mapped this would have to
be manually updated from within the DSP driver itself, which is hardly
very idiomatic.
This patch switches to using the cumulative values to calculate the
available buffer space because these are already gracefully passed out
of the DSP driver to the compress core and otherwise should be
functionally equivalent.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is my example conversion of a few existing mmap users. The pcm
mmap case is one of the more straightforward ones.
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
Use the generic dmaengine PCM driver instead of a custom implemention. There is
a minor functional change, the ux500 PCM driver did not preallocate the audio
buffer, while the generic dmaengine PCM driver will do this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.
That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.
The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.
ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.
This patch adds support for this by adding a boolean flag to the
audio format struct.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.
The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.
To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.
The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When pin default configs are overridden via patch option, these are
evaluated before fixups are applied. Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.
We may reorder the execution, but an easier fix is simply to disable
this sanity check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The flag bus->shutdown implies that the control elements might have
been already destroyed. When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.
This patch just adds a check of the flag in the caller side for
avoiding such a crash.
Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily. So let it be for a while.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware revision of the codec is based at 0x40. Subtract that
before convering to ASCII. The same as it is done for 98095.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
This patch adds a playback and capture streams to the dummy codec DAI
configuration. Most permissive set of sampling rates and formats is used.
This patch is needed for playback and capturing on a codec-less systems,
as otherwise the PCM device nodes are not even created.
Signed-off-by: Stas Sergeev <stsp@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows us to access the DAI DMA data when we create the PCM. We'll use
this when converting imx to generic DMA engine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunately there are still quite a few platforms with a dmaengine driver
which do not support reporting the number of bytes left to transfer. If we want
to support these platforms in the generic dmaengine PCM driver we have.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the generic dmaengine PCM driver instead of a custom implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for platforms which don't use devicetree yet or have to optionally
support a non-devicetree way to request the DMA channel. The patch adds the
compat_request_channel and compat_filter_fn callbacks to the
snd_dmaengine_pcm_config struct. If the compat_request_channel is implemented it
will be used to request the DMA channel. If not dma_request_channel with
compat_filter_fn as the filter function will be used to request the channel.
The patch also exports the snd_dmaengine_pcm_request_chan() function, since
compat platforms will want to use it to request their DMA channel.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a generic dmaengine PCM driver. It builds on top of the
dmaengine PCM library and adds the missing pieces like DMA channel management,
buffer management and channel configuration. It will be able to replace the
majority of the existing platform specific dmaengine based PCM drivers.
Devicetree is used to map the DMA channels to the PCM device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_{add,remove}_platform are similar to snd_soc_register_platform and
snd_soc_unregister_platform with the difference that they won't allocate and
free the snd_soc_platform structure.
Also add snd_soc_lookup_platform which looks up a platform by the device it has
been registered for.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the dmaengine PCM library to allow the DMA channel to be requested
before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA
channel instead of a filter function and filter parameter as its parameters.
snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows
a dmaengine based PCM driver to request its channels before the substream is
opened.
The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan()
and snd_dmaengine_pcm_close_release_chan(), which have the same signature and
behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new
variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are
updated to use snd_dmaengine_pcm_open_request_chan() and
snd_dmaengine_pcm_close_release_chan().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When graphics initializes the HDMI chip, sometimes this leads to
pins going into D3 and right channel being muted. If the audio driver
finishes initialization before the graphic driver does, this situation
becomes permanent.
This is a workaround that checks for this situation and corrects it on
playback prepare. It has been verified working on at least one machine.
BugLink: https://bugs.launchpad.net/bugs/1167270
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default. Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add definitions for AC97 control register.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not. But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.
Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing. This leaves the aamix path even though the
loopback control is turned off.
This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capture, the delay through the codec contributes to the time stamp
of the sample recorded at the A to D. Rename the codec time stamp
function appropriately.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices. However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions. With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.
Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.
Commit f5f165418c (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.
To work around all this, just disable autopm for all USB MIDI devices.
Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this patch, a TRRS headset mic cannot be successfully detected
on the Asus X101CH, and we can also distinguish between headphone
and headset automatically.
Buglink: https://bugs.launchpad.net/bugs/1169138
Co-authored-by: Kailang <kailang@realtek.com>
Tested-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some machines, there is a headset jack that can support both
headphone, headsets (of both CTIA and OMTP type) and mic-in.
On other machines, the headset jack supports headphone, headsets
(both CTIA and OMTP), but not mic-in.
This patch implements that functionality as different capture sources.
Buglink: https://bugs.launchpad.net/bugs/1169143
Tested-by: David Chen <david.chen@canonical.com>
Co-authored-by: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.
Userspace expected: L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1
Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.
Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.
Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit b2ca78717c (ARM: S3C24XX: make gta02.h local) already replaced
the GTA02_GPIO_* constants in neo1973-wm8753.c but forgot to remove the
inclusion of mach/gta02.h before moving the file out of mach/.
Signed-off-by: Heiko Stuebner <heiko@sntech.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few updates, more than I'd like, fixing some relatively small issues
but mostly driver specific ones. Nothing wildly exciting so if it
doesn't make v3.9 it won't be the end of the world but it'd be nice.
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Merge tag 'asoc-v3.9-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.9
A few updates, more than I'd like, fixing some relatively small issues
but mostly driver specific ones. Nothing wildly exciting so if it
doesn't make v3.9 it won't be the end of the world but it'd be nice.
The plat/regs-iis.h and plat/regs-ac97.h files in the samsung platform
are only needed by the ASoC drivers, so they can be moved into the same
directory, as one more step towards a multiplatform build.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Firmwares may provide some firmware wide configuration regions which can
be configured by the coefficient files using the firmware ID as the
algorithm ID, include these in the algorithm list.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The idma_reg_addr_init function is used by the samsung i2s driver,
which can be a loadable module, so we have to export this function.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The second argument to the module_device_table macro must be the
name of the device id array. In the samsung i2s driver, there
was a small typo, resulting in a build error when building it
as a loadable module.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With multiplatform kernels, we cannot use hardwired IRQ
numbers in device drivers. This changes the idma driver
to use a proper resource, like all other drivers do.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I never liked that we move our speaker and hp pins to line out
if there are not any line outs; but now that we do,
add some convenience functions to find hp and speaker pins even
if they have been moved.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows a specific mic to get the "Headphone Mic" name, in addition
to the existing "Headset Mic" name.
Also, it allows for a special mark: if the sequence number is set
to 0xc, that's an indication to prefer it for headset mic, and if it's
set to 0xd, that's an indication to prefer it for headphone mic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only part of proc_dir_entry the code outside of fs/proc
really cares about is PDE(inode)->data. Provide a helper
for that; static inline for now, eventually will be moved
to fs/proc, along with the knowledge of struct proc_dir_entry
layout.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
The Charge Pump needs the DSP clock to work properly, without it the
bypass to HP/LINEOUT is not working properly. This requirement is not
mentioned in the datasheet but has been confirmed by Mark Brown from
Wolfson.
Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
