On Qualcomm platforms, specifically with SLIMbus interfaced codecs,
the codec slim channel numbers are passed to DSP while configuring
the slim audio path. Having get_channel_map() would allow dais to
share such information across multiple dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
We've got many code additions at this cycle as a result of quite a few
new drivers. Below are highlights:
Core stuff:
- Fix the long-standing issue with the device registration order;
the control device is now registered at last
- PCM locking code cleanups for RT kernels
- Fixes for possible races in ALSA timer resolution accesses
- TLV offset definitions in uapi
ASoC:
- Many fixes for the topology stuff, including fixes for v4 ABI
compatibility
- Lots of cleanups / quirks for Intel platforms based on Realtek
CODECs
- Continued componentization works, removing legacy CODEC stuff
- Conversion of OMAP DMA to the new, more standard SDMA-PCM driver
- Fixes and updates to Cirrus Logic SoC drivers
- New Qualcomm DSP support
- New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306 and
RT5668 and TI TSCS454
HD-audio:
- Finally better support for some CA0132 boards, allowing Windows
firmware
- HP Spectre x360 support along with a bulk of COEF stuff
- Blacklisting power save default some known boards reported on Fedora
USB-audio:
- Continued improvements on UAC3 support; now BADD is supported
- Fixes / improvements for Dell WD15 dock
- Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co
Others:
- New Xen sound frontend driver support
- Cache implementation and other improvements for FireWire DICE
- Conversions to octal permissions in allover places
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Merge tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've got many code additions at this cycle as a result of quite a few
new drivers. Below are highlights:
Core stuff:
- Fix the long-standing issue with the device registration order; the
control device is now registered at last
- PCM locking code cleanups for RT kernels
- Fixes for possible races in ALSA timer resolution accesses
- TLV offset definitions in uapi
ASoC:
- Many fixes for the topology stuff, including fixes for v4 ABI
compatibility
- Lots of cleanups / quirks for Intel platforms based on Realtek
CODECs
- Continued componentization works, removing legacy CODEC stuff
- Conversion of OMAP DMA to the new, more standard SDMA-PCM driver
- Fixes and updates to Cirrus Logic SoC drivers
- New Qualcomm DSP support
- New drivers for Analog SSM2305, Atmel I2S controllers, Mediatek
MT6351, MT6797 and MT7622, Qualcomm DSPs, Realtek RT1305, RT1306
and RT5668 and TI TSCS454
HD-audio:
- Finally better support for some CA0132 boards, allowing Windows
firmware
- HP Spectre x360 support along with a bulk of COEF stuff
- Blacklisting power save default some known boards reported on
Fedora
USB-audio:
- Continued improvements on UAC3 support; now BADD is supported
- Fixes / improvements for Dell WD15 dock
- Allow DMA coherent pages for PCM buffers for ARCH, MIPS & co
Others:
- New Xen sound frontend driver support
- Cache implementation and other improvements for FireWire DICE
- Conversions to octal permissions in allover places"
* tag 'sound-4.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (386 commits)
ASoC: dapm: delete dapm_kcontrol_data paths list before freeing it
ALSA: usb-audio: remove redundant check on err
ASoC: topology: Move skl-tplg-interface.h to uapi
ASoC: topology: Move v4 manifest header data structures to uapi
ASoC: topology: Improve backwards compatibility with v4 topology files
ALSA: pci/hda: Remove unused, broken, header file
ASoC: TSCS454: Add Support
ASoC: Intel: kbl: Move codec sysclk config to codec_init function
ASoC: simple-card: set cpu dai clk in hw_params
ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
ALSA: oxygen: use match_string() helper
ASoC: dapm: use match_string() helper
ASoC: max98095: use match_string() helper
ASoC: max98088: use match_string() helper
ASoC: Intel: bytcr_rt5651: Set card long_name based on quirks
ASoC: mt6797-mt6351: add hostless phone call path
ASoC: mt6797: add Hostless DAI
ASoC: mt6797: add PCM interface
ASoC: mediatek: export mtk-afe symbols as needed
ASoC: codecs: PCM1789: include gpio/consumer.h
...
SoundWire stream needs to be propagated to all the DAIs(cpu, codec).
So, add a snd_soc_dai_set_sdw_stream() API for the same.
Signed-off-by: Shreyas NC <shreyas.nc@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Now no one is using Codec related code.
Let's remove all
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment playback_active and capture_active are using only 1 bit so
the maximum active count is 1.
However, snd_soc_runtime_activate() may be called several time on the
same dai. This happens when a dai is part of several dai_links. It is
often the case for "snd-soc-dummy-dai".
This is a problem if snd_soc_runtime_activate() is called an even number
of times on a dai. In this case the active count overflow back to 0. As
consequence, ASoC functions, such as soc_dpcm_runtime_update(), won't run
correctly.
Storing these usage counts on plain 'unsigned int' solves the problem.
Fixes: f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC has some duplicate parameter.
snd_soc_dai::symmetric_xxx are one of them.
commit f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component
Support") moved "symmetric_xxx" flags from snd_soc_dai to
snd_soc_dai_driver. No one is using snd_soc_dai::symmetric_xxx now.
Let's remove it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep
20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not
allow 3-byte accesses (including DMA) so a 4-byte (more conventional)
format is needed for it.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During probe, DAIs can need to perform some actions that requests
the knowledge of the pcm runtime handle.
The callback is called during DAIs linking, after PCM device creation.
For instance this can be used to add relationship between a DAI pcm
control and the pcm device.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC Compress ops have only platform ops and no DAI ops unlike PCM device
where we have both platform ops as well as DAI ops.
So add compress dai ops and add this new structure to the ASoC core to make
compressed devices a first class ASoC citizen
Again like PCM ops, drivers are free to implement either or both of
these ops based on device needs.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can reduce struct size in certain environment.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Define DAI physical PCM data formats for user space, so users can
configure the formats of backends by topology (e.g. the DAI format
to set on backend link init).
The kernel will also refer to these formats.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Define API snd_soc_register_dai() to add a DAI dynamically and
create the DAI widgets. Topology can use this API to register DAIs
when probing a component with topology info. These DAIs's playback
& capture widgets will be freed when the sound card is unregistered
and the DAIs will be freed when cleaning up the component.
And a dobj is embedded into the struct snd_soc_dai_driver. Topology
can use the dobj to find the DAI drivers created by it and free them
when the topology component is removed.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently there is no clear definition of what FSYNC polarity is.
Different drivers use its own definition of what is "normal" and what is
"inverted" fsync. This leads to compatibility problems between drivers.
For example TegraX1 driver assumes that DSP-A format with frames
starting at rising FSYNC edge has "inverted" polarity,
while RT5677 assumes it is "normal" polarity.
Explicitly specify meaning of BCLK/FSYNC polarity to avoid future
compatibility problems.
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't always need soc-compress in soc, here add a config item
SND_SOC_COMPRESS, when nobody select it, the soc-compress will
not be compiled.
Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers
that needed soc-compress.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Setting the ac97_control field on a CPU DAI tells the ASoC core that this
DAI in addition to audio data also transports control data to the CODEC.
This causes the core to suspend the DAI after the CODEC and resume it before
the CODEC so communication to the CODEC is still possible. This is not
necessarily something that is specific to AC'97 and can be used by other
buses with the same requirement. This patch renames the flag from
ac97_control to bus_control to make this explicit.
While we are at it also change the type from int to bool.
The following semantich patch was used for automatic conversion of the
drivers:
// <smpl>
@@
identifier drv;
@@
struct snd_soc_dai_driver drv = {
- .ac97_control
+ .bus_control
=
- 1
+ true
};
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This has no users since commit f0fba2ad1b ("ASoC: multi-component - ASoC
Multi-Component Support") which was almost 5 years ago. Given that this runs
after CODEC probe functions have been run it also doesn't seem to be that
useful.
So drop it altogether to make the code simpler.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The card field of the snd_soc_dai field is very rarely used. We can use
dai->component->card instead and remove the card field from the snd_soc_dai
struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Typically a DAI does not need direct access to the platform. Currently the
only user of this field is in a platform driver where we have a more direct
way of getting a pointer to the platform. This patch updates the driver to
use the more direct way and then removes the platform field from the
snd_soc_dai struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAI link assumes a one to one mapping between CPU DAI and CODEC. In
some cases, the same CPU DAI can be connected to several codecs.
This is the case for example, if you connect two mono codecs to the
same I2S link in order to have a stereo card.
The current ASoC implementation does not allow such setup.
Add support for DAI links composed of a single CPU DAI and multiple
CODECs. Sound cards have to pass the CODECs array in the corresponding
DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in
this array is described in the same manner single CODEC DAIs are
(either DT/OF node or codec_name).
Multi-codec links are not supported in the case of CODEC to CODEC links.
Just print a warning if it happens.
Based on an original code done by Misael.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-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 was initially removed in commit 6423c1875 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad1 ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support") added
a per card list that keeps track of all the DAIs that have been registered with
the card, but the list has never been used. This patch removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The 'of_' is not appropriate here for there hasn't any DT parsing.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Keep track of which component registered a DAI. We'll need this as
componentization progresses.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some CPU/CODEC DAI devices the TDM slot infomation maybe needed. This
patch adds the slot parsing from DT supports.
TDM slot properties:
dai-tdm-slot-num : Number of slots in use.
dai-tdm-slot-width : Width in bits for each slot.
For instance:
dai-tdm-slot-num = <2>;
dai-tdm-slot-width = <8>;
And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
to specify a explicit mapping of the channels and the slots. If it's absent
the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
tx and rx masks.
For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
for an active slot as default, and the default active bits are at the LSB of
the masks.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI.
This is for internal DAPM usage only and is used to determine whether to
insert a DAI link connection into the DAPM graph.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.
In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.
Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.
Thus, this patch applys symmetry for channels and sample bits.
Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow DMA data to be set at probe time for devices that can do that,
avoiding the need to do it every time we start a stream and supporting
non-DT dmaengine users using the helpers.
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a comment to the trigger function in snd_soc_dai_ops struct about
possible command sequences.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some codec drivers when running in slave mode require that BCLK to sample rate ratio
is explicitly set by the machine driver as it may not be exactly rate * frame size.
Extend the DAI API by adding :-
int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 << 4),
but gated clock should be default settings (= 0).
This patch fixup SND_SOC_DAIFMT_GATED as (0 << 4).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 << 8),
but normal bit clock / normal frame should be
default settings (= 0).
This patch fixup SND_SOC_DAIFMT_NB_NF as (0 << 8).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Here we update the asoc structures to add compress stream definations
First the struct snd_soc_dai_driver adds a new member to indicate if the dai is
compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in
the struct snd_soc_dai_link. This is to be used for machine driver to perform
any opertaions required for setting up compressed audio streams
next is the compressed data operations, they are added using struct
snd_compr_ops in the struct snd_soc_platform_driver.
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices with many DAIs are becoming more and more common, and generally
the more modern devices have consistent register layouts between DAIs.
Rather than have drivers open code lookups based on the DAI ID or cause
uglification in UI by having register addresses for IDs provide a base
address field they can use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.
A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
