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>
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>
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>
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>
