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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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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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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-dapm
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soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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There is no need to use a normal per-CPU workqueue for delayed power downs
as they're not timing or performance critical and waking up a core for them
would defeat some of the point.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Viresh Kumar <viresh.kumar@linaro.org>
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Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.
Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.
Reported-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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When declaring playback and capture capabilities check for both CODEC
side and CPU side support rather than only checking for CODEC side
support. While it is unusual some CPUs do have unidirectional DAIs.
Reported-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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We use the same code to initialize the runtime pcm based on the
snd_soc_pcm_stream struct on both the playback and capture path. Factor this
code into a helper function to make things a bit more tidy.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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snd_soc_set_runtime_hwparams() is the only PCM related function that lives in
soc-core.c. All other PCM related functions live in soc-pcm.c, so move
snd_soc_set_runtime_hwparams() over as well for a bit more consistency.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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I've removed several unreachable returns.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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