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In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples
are stored and transferred in memory on 32bits (lower 3 bytes of the 4
bytes).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Correct the hw_params callback to configure the codec correctly in case of
S24_3LE format since in case of S24_3LE the codec has been configured to
16bit format mode.
S24_LE is not defined as supported format for the codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current DVC can be enabled only when playback,
but, this came from misunderstanding.
It is not correct.
DVC <-> DMA relationship is...
Playback: MEM -> DMAC -> SRC -> DVC -> DMACp -> SSI
Capture: SSI -> DMACp -> SRC -> DVC -> DMAC -> MEM
DVC can be used for both Playback/Capture
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Sound data needs to be sent to R-Car sound SSI when playback.
But, there are 2 interfaces for it.
1st is SSITDR/SSIRDR which are mapped on SSI.
2nd is SSIn_BUSIF which are mapped on SSIU.
2nd SSIn_BUSIF is used when DMA transfer,
and it is always used if sound data came from via SRC.
But, we can use it when SSI+DMA case too.
(Current driver is assuming 1st SSITDR/SSIRDR for it)
2nd SSIn_BUSIF can be used as FIFO.
This is very helpful/useful for SSI+DMA.
But DMA address / DMA ID are not same between 1st/2nd cases.
This patch care about these settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current FSI driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsi PIO/DMA handler are using each own pointer update method,
but these can be share.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current fsi driver is using SNDRV_DMA_TYPE_CONTINUOUS
for snd_pcm_lib_preallocate_pages_for_all().
But, it came from original dma-sh7760.c,
and no longer needed.
This patch exchange its parameter, and removed
original dma mapping and un-needed
dma_sync_single_xxx() from driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Avoid creating duplicate directories by prefixing codecs and platforms
with their separate identifiers. This avoids snd-soc-dummy (which can
appear both as a dummy platform and a dummy codec on the same card)
from clashing.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
The similar fixup as T440 is needed for supporting the dock on T540.
Reported-by: Jim Minter <jminter@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another quirk to make the headset mic work on some new Dell machines.
Cc: Hui Wang <hui.wang@canonical.com>
BugLink: https://bugs.launchpad.net/bugs/1297581
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Intel Haswell/Broadwell display HD-A controller, the 24MHz HD-A link BCLK
is converted from Core Display Clock (CDCLK): BCLK = CDCLK * M / N
And there are two registers EM4 and EM5 to program M, N value respectively.
The EM4/EM5 values will be lost and when the display power well is disabled.
BIOS programs CDCLK selected by OEM and EM4/EM5, but BIOS has no idea about
display power well on/off at runtime. So the M/N can be wrong if non-default
CDCLK is used when the audio controller resumes, which results in an invalid
BCLK and abnormal audio playback rate. So this patch saves and restores valid
M/N values on controller suspend/resume.
And 'struct hda_intel' is defined to contain standard HD-A 'struct azx' and
Intel specific fields, as Takashi suggested.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a USB-audio device is disconnected while PCM is still running, we
still see some race: the disconnect callback calls
snd_usb_endpoint_free() that calls release_urbs() and then kfree()
while a PCM stream would be closed at the same time and calls
stop_endpoints() that leads to wait_clear_urbs(). That is, the EP
object might be deallocated while a PCM stream is syncing with
wait_clear_urbs() with the same EP.
Basically calling multiple wait_clear_urbs() would work fine, also
calling wait_clear_urbs() and release_urbs() would work, too, as
wait_clear_urbs() just reads some fields in ep. The problem is the
succeeding kfree() in snd_pcm_endpoint_free().
This patch moves out the EP deallocation into the later point, the
destructor callback. At this stage, all PCMs must have been already
closed, so it's safe to free the objects.
Reported-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Spectre 13 has the IDT 92HD95 codec, and BIOS seems to set the
default high-pass filter in some "safer" range, which results in the
very soft tone from the built-in speakers in contrast to Windows.
Also, the mute LED control is missing, since 92HD95 codec still has no
HP-specific fixups for GPIO setups.
This patch adds these missing features: the HPF is adjusted by the
vendor-specific verb, and the LED is set up from a DMI string (but
with the default polarity = 0 assumption due to the incomplete BIOS on
the given machine).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=74841
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out there is no need to enable microphone detection in MAX98090
codec. Headset microphone is anyway detected by a GPIO signal from another
chip and headset button presses cannot be detected either because a signal
needed for it is not connected.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Pass actual jack type bitmask to snd_soc_jack_new() in order to report
also microphone detections and not only headphone. While at it change also
jack name and pass also SND_JACK_LINEOUT type.
Reported-by: Jin Yao <yao.jin@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Headset jack has only mono microphone input.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Mic detect GPIO is active low when headset microphone is detected. Found
both by debugging and checking the schematics.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it
between input pin "IN34" and "Headset Mic".
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is cosmetical - it makes the pin quirk table look better.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is cosmetical - it makes the new pin quirk table look better.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two bug reporters with Dell XPS 15 report that they need to use the
dell-headset-multi model to get the headset mic working.
The two bug reporters have different PCI SSID (1028:05fd and 1028:05fe)
but this pin quirk matches both.
BugLink: https://bugs.launchpad.net/bugs/1331915
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to call the proper init function in case it has been
overridden, as it might restore things that the generic routing
doesn't know anything about. E.g. AMD cards have special verbs
that need resetting.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=77901
Fixes: 5a61358433b1 ('ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support')
Signed-off-by: Pierre Ossman <pierre@ossman.eu>
Cc: <stable@vger.kernel.org> [v3.13+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A recent refactoring broke the possibility to manually specify
model name as a module parameter. This patch restores the desired
functionality.
Fixes: c21c8cf77f47 ('ALSA: hda - Add fixup_forced flag')
Reported-by: Kent Baxley <kent.baxley@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs
is now unused can be removed. New code that wants to get the platform for a
widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or
snd_soc_dapm_to_platform(dapm).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI
widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC:
core: Only add platform DAI widgets once."). Now that we have a per component
DAPM context it is unlikely that we'll need the DAI DAPM context again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds stream_event() and seq_notifier() callbacks similar to those
found in the snd_soc_codec_driver and snd_soc_platform driver struct to the
snd_soc_component_driver struct. This is meant to unify the handling of these
callbacks across different types of components and will eventually allow their
removal from the CODEC and platfrom driver structs.
The new callbacks are slightly different from the old ones in that they take a
snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was
done since otherwise casting from the DAPM context to the component would
typically be the first thing to do in the callback. And the interface becomes
slightly cleaner by passing a snd_soc_component to all callbacks in the
snd_soc_component_driver struct.
The patch also already removes the stream_event() callback from the
snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently
unused.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch
it over to using the snd_soc_component DAPM context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds full DAPM support at the component level. Previously there was
only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor
MUXs) for platforms. Having DAPM support at the component level will allow all
types of components to use DAPM and also help in consolidating the DAPM support
between CODECs and platforms.
Since the DAPM context is directly embedded into the snd_soc_codec and
snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of
drivers moving the field just right now is not possible without causing code
churn. The approach this patch takes is to add two new fields to the component
struct. One field which is the pointer to the actual DAPM context used by the
component and one DAPM context that will be used as the default if no other
context was specified. For CODECs and platforms the pointer is initialized to
point to the CODEC or platform DAPM context. All generic code when referencing
a component's DAPM struct will go via the pointer. This will make it possible to
eventually seamlessly move the DAPM context from snd_soc_codec and
snd_soc_platform struct over once all direct references have been eliminated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code directly looks at the CODEC driver struct to get a
handle to the set_bias_level() callback. This patch adds a new set_bias_level()
callback to the DAPM context struct. The DAPM code will use this new callback
instead of the CODEC callback. For CODECs the new callback is set up to call the
CODEC specific set_bias_level callback(). Not looking directly at the CODEC
driver struct will allow non CODEC DAPM contexts to implement a set_bias_level()
callback.
This is also similar to how the seq_notifier() and stream_event() callbacks are
currently handled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently only pins in CODEC DAPM contexts are automatically marked as
non-connected if the card has the fully_routed flag set. This makes sense since
widgets which qualify for auto-disconnection are only found in CODEC DAPM
contexts. But with componentisation this is going to change, so consider all
widgets for auto-disconnection.
Also it is probably faster to walk the widgets list only once rather than once
for each CODEC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Split snd_soc_component_register() into snd_soc_component_initialize() and
snd_soc_component_add(). Using a 2-stage registration approach has the advantage
that it is possible to modify the component after it has been initialized, but
before it is made visible to the system. This e.g. allows CODECs or platforms to
overwrite some of the default settings made in snd_soc_component_initialize().
Similar snd_soc_component_unregister() is split into two steps as well,
snd_soc_component_delete(), which removes the component from the system, and
snd_soc_component_cleanup(), which frees all the resources allocated by the
component.
Furthermore this patch makes sure that if a component is visible on two list
(e.g. the component list and the CODEC list) it is added or removed to both
lists atomically.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The component struct already has a name and id field which are initialized to
the same values as the same fields in the CODEC and platform structs. So remove
them from the CODEC and platform structs and used the ones from the component
struct instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move the name_prefix from the CODEC struct to the component struct. This will
eventually allow to specify prefixes for all types of components. It is also
necessary to make the DAPM code component type independent (i.e. a DAPM context
does not need to know whether it belongs to a CODEC or a platform or something
else).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a
enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE
and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same
value the code worked fine even though it was using the wrong type.
Fixes the following warning from sparse:
sound/soc/sh/rcar/core.c:227:49: warning: mixing different enum types
sound/soc/sh/rcar/core.c:227:49: int enum dma_data_direction versus
sound/soc/sh/rcar/core.c:227:49: int enum dma_transfer_direction
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When SSC work as master, it will generate the frame sync signal.
On old SoCs, it only supports frame sync length less or equal to
16bits, on newer SoCs, it supports frame sync length extension,
which can support frame size larger than 16 bits.
So, add this to make it supports playback 24/32 bits audio clips.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
On the wm5110 CODEC both the left and right channel must be powered
when an output is being used as a mono output, although no audio is
routed to the right output channel. This patch adds additional DAPM
routes to link the right channel to the left in the case where an output
is marked as mono. Audio must always be brought in on the left channel
for mono operation.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need to release dapm widget list after dpcm_path_get in
soc_dpcm_runtime_update. otherwise, there will be potential memory
leak. add dpcm_path_put to fix it.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Passing unsigned int pointers as u32 ponters may be dangerous on 64-bit
system.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes build with SND_DAVINCI_SOC or SND_OMAP_SOC alone and adds build
dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
max98090.c doesn't free the threaded interrupt it requests. This causes
an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is
unloaded.
Fix this by requesting the interrupt by using devm_request_threaded_irq().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Stable <stable@vger.kernel.org> # 3.10+
Signed-off-by: Mark Brown <broonie@linaro.org>
We now have a generic helper function to cast from a DAPM context to a CODEC.
Make use of it in the places which previously open-coded it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Document the newly added regulators to the DT binding document.
Also, "static const char const *x" is not identical to "static const
char * const x", which sparse now complains about. Fix it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The chip has two power supplies, VA and VDD. Enable them both as long
as the codec is in use.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
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Merge tag 'asoc-v3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.16
Quite a few build coverage fixes in here among the usual small driver
fixes includling the sigmadsp change from Lars - moving the driver to
separate modules per bus (which is basically just code motion) avoids
issues with some combinations of buses being enabled.
The ALSA control code expects that the range of assigned indices to a control is
continuous and does not overflow. Currently there are no checks to enforce this.
If a control with a overflowing index range is created that control becomes
effectively inaccessible and unremovable since snd_ctl_find_id() will not be
able to find it. This patch adds a check that makes sure that controls with a
overflowing index range can not be created.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
