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Add a function helper to factorize the hw_params code.
Suggested by Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Benoit Cousson <bcousson@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>
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Add multicodec support in soc-pcm.c
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>
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Refactor the function to facilitate the migration to
multiple codecs.
Fix a trailing space in the header as well.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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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>
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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
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'asoc/topic/jz4740', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/mc13783' and 'asoc/topic/multicodec' into asoc-next
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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>
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The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be
SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Machine specific trigger callback allows to do final stream start/stop
related operations in a machine driver after setting up the codec, DMA and
DAI.
One example could be clock management for linked streams case where machine
driver can start/stop synchronously the linked streams.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Stefan Roese <sr@denx.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Replace rtd_get_codec_widget() and rtd_get_cpu_widget() by a simple
dai_get_widget() in preparation for DAI-multicodec support, per Lars
suggestion.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
# gpg: Signature made Wed 12 Mar 2014 23:05:45 GMT using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
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The changes in "ASoC: pcm: free path list before exiting from error
conditions" actually introduced both double frees (in case where the
path list was allocated but empty) and frees of unallocated memory (in
cases where the error being handled was -ENOMEM. Drop the commit for
now.
Fixes: e4ad1accb (ASoC: pcm: free path list before exiting from error conditions)
Reported-by: Ben Hutchings <ben@decadent.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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In preparation for componentization move the ignore_pmdown_time field from the
snd_soc_codec struct to the snd_soc_component struct. Set it to true for non
CODEC components for now.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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There is no reason why active count tracking should only be done for CODECs but
not for other components. Moving the active count from the snd_soc_codec struct
to the snd_soc_component struct reduces the differences between CODECs and other
components and will eventually allow component to component DAI links (Which is
a prerequisite for converting CODECs to components).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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For CODEC to CODEC links we need to make sure to also manage the 'active' field
of the cpu_dai CODEC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We have the same code that increments and decrements the active field of the
various PCM runtime components (all with the same bugs). Factor this out into
common helper functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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For CODEC to CODEC links we should only immediately power down if both CODECs
are configured to ignore the power down delay. Factor the logic for this
into a helper function that can be used for both compressed and normal PCMs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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dpcm_path_get() allocates dynamic memory to hold path list.
Corresponding dpcm_path_put() must be called to free the memory.
dpcm_path_put() is not called under several error conditions.
This leads to memory leak.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Instead of open-coding the intersecting of two rate masks (and getting slightly
wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect()
helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll
end up with the rate_max field of the runtime hardware set to 0. (Note that it
is still possible for the components to constrain the supported sample rates
using other methods, e.g. setting a list constraint) If rate_max is 0 this means
that the sound card doesn't support any rates at all, which is not the desired
result. So initialize rate_max to UINT_MAX. For symmetry reasons also set
rate_min to 0.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Linux 3.13-rc3
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Some DMA cores might add additional restrictions on which in memory audio
formats can be supported by the compound sound card. If the PCM component
specifies a set of formats it supports (by setting the formats field to non 0)
take these into account when calculating the format set for the compound sound
card.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).
Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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When the platform driver has no ops, the platform function
bespoke_trigger() is no more called.
The problem was introduced by the commit c5914b0aaea6494aaa9e415cbd32f8b7eb604af0
"ASoC: pcm: Check for ops before deferencing them"
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Allow PCMs that do not impose any restrictions on the supported formats to set
the formats field to 0, Instead of assuming that this means that the PCM does
not support any formats (which doesn't make much sense), assume that it supports
all formats. This brings the behavior of DPCM closer to that of non-DPCM.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We have the same code for initializing the runtime pcm on both the playback and
the capture path. Factor this out into a common helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Conflicts (Trivial add/delete):
sound/soc/soc-pcm.c
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The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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If there are symmetry constraints between the playback and the capture channel
set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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snd_pcm_limit_hw_rates() will initialize the minimum and maximum sample rate for
the PCM stream based on the rates specified in the rates field. Since we call
snd_pcm_limit_hw_rates() after soc_pcm_init_runtime_hw() it will essentially
overwrite the min and max rate set in soc_pcm_init_runtime_hw(). This may cause
the minimum or maximum rate to be set to a value outside the range of one of the
components if one of the components sets either SNDRV_PCM_RATE_CONTINUOUS or
SNDRV_PCM_RATE_KNOT and the other component specified a discrete rate via
SNDRV_PCM_RATE_[0-9]* that is outside of the first component's rate range. To
fix this first calculate the minimum and maximum rates using
snd_pcm_limit_hw_rates() and then on top of that apply the contraints specified
in the snd_soc_pcm_stream structs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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In order to make sure that the sample rate is in the supported range of both
components the maximum rate of the card should be the minimum of the maximum
rate of each components. There is one special case to consider though, if
max_rate is set to 0 this means there is no maximum specified, so use
min_not_zero() macro which will give use the desired result.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Takashi iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We're now applying soc_hw_params_symmetry() to reject unmatched parameters
while we clear parameters in soc_pcm_close(). So here's a use case might be
broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav
In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free()
->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we
only clear parameters in soc_pcm_close(). The parameters would be remained
in the system even if the playback of 44100.wav is finished.
Thus, this patch is trying to move parameters cleaning into hw_free() so that
the system can continue to serve this kind of use case.
Also, since we set them in hw_params(), it should be better to clear them in
hw_free() for symmetry.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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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>
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It's quite popular that more drivers are using pinctrl PM, for example:
(Documentation/devicetree/bindings/arm/primecell.txt). Just like what
runtime PM does, it would deactivate and activate pin group depending
on whether it's being used or not.
And this pinctrl PM might be also beneficial to cpu dai drivers because
they might have actual pinctrl so as to sleep their pins and wake them
up as needed.
To achieve this goal, this patch sets pins to the default state during
resume or startup; While during suspend and shutdown, it would set pins
to the sleep state.
As pinctrl PM would return zero if there is no such pinctrl sleep state
settings, this patch would not break current ASoC subsystem directly.
[ However, there is still an exception that the patch can not handle,
that is, when cpu dai driver does not have pinctrl property but another
device has it. (The AUDMUX <-> SSI on Freescale i.MX6 series for example.
SSI as a cpu dai doesn't contain pinctrl property while AUDMUX, an Audio
Multiplexer, has it). In this case, this kind of cpu dai driver needs to
find a way to obtain the pinctrl property as its own, by moving property
from AUDMUX to SSI, or creating a pins link/dependency between these two
devices, or using a more decent way after we figure it out. ]
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Avoid oopsing if there is no backend stream associated with a front end
stream.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Ensure that we always check that an ops structure is present before we
try to use it, improving the robustness of the system.
Reported-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
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dev_ prints are already prefixed by ": " before format string so there is no
need for extra spaces.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.
The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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