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>
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>
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>
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>
