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>
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>
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.
