Dan Carpenter [Fri, 7 Sep 2018 19:40:33 +0000 (22:40 +0300)]
ASoC: dapm: Fix a couple uninitialized ret variables
Smatch complains that these variables could be uninitialized. The first
one in snd_soc_dai_link_event() is probably a false positive, because
probably we know the lists are not empty. I would normally ignore the
warning, but GCC complains here as well so I just silenced the warning.
The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it
leads to a bogus dereference in the caller.
Fixes: 3bbf5d34fd4a ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked") Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Commit ca917f9fe1a0fab added use of usleep_range() but not
the corresponding "include <linux/delay.h>". The result is
with Chrome OS won't build because warnings are forced
to be errors:
mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration]
usleep_range(10000, 11000);
^
Including delay.h "fixes" this.
Signed-off-by: Grant Grundler <grundler@chromium.org> Reviewed-by: Benson Leung <bleung@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Thu, 6 Sep 2018 16:41:55 +0000 (17:41 +0100)]
ASoC: dapm: Avoid uninitialised variable warning
Commit 4a75aae17b2a ("ASoC: dapm: Add support for multi-CODEC
CODEC to CODEC links") adds loops that iterate over multiple
CODECs in snd_soc_dai_link_event. This also introduced a compiler
warning for a potentially uninitialised variable in the case
no CODECs are present. This should never be the case as the
DAI link must by definition contain at least 1 CODEC however
probably best to avoid the compiler warning by initialising ret
to zero.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Colin Ian King [Thu, 6 Sep 2018 10:41:52 +0000 (11:41 +0100)]
ASoC: sgtl5000: avoid division by zero if lo_vag is zero
In the case where lo_vag <= SGTL5000_LINE_OUT_GND_BASE, lo_vag
is set to zero and later vol_quot is computed by dividing by
lo_vag causing a division by zero error. Fix this by avoiding
a zero division and set vol_quot to zero in this specific case
so that the lowest setting for i is correctly set.
Signed-off-by: Colin Ian King <colin.king@canonical.com> Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd driver sometimes want to know which address is used when debugging.
But it will indicate "(____ptrval____)" if it used "%p" on dev_dbg().
Let's use "%pa" or "%px" for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
commit 8c9d75033340 ("ASoC: rsnd: ssiu: Support BUSIF
other than BUSIF0") added new SSIU registers.
But it is using white-space for it.
This patch fixup it to use tab.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Wed, 5 Sep 2018 14:21:02 +0000 (15:21 +0100)]
ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime
Larger CODECs may contain many several hundred widgets and which set of
parameters is selected only needs to be recorded on a per DAI basis. As
such move the selected CODEC to CODEC link params to be stored in the
runtime rather than the DAPM widget, to save some memory.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Wed, 5 Sep 2018 14:21:01 +0000 (15:21 +0100)]
ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links
Currently multi-CODEC is not supported on CODEC to CODEC links.
There are common applications where this would be useful, such
as connecting two mono amplifiers to an audio CODEC. Adding
support simply requires an update of snd_soc_dai_link_event
to loop over the attached CODEC DAIs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Wed, 5 Sep 2018 14:21:00 +0000 (15:21 +0100)]
ASoC: dapm: Move connection of CODEC to CODEC DAIs
Currently, snd_soc_dapm_connect_dai_link_widgets connects up the routes
representing normal DAIs, however CODEC to CODEC links are hooked up
through separate infrastructure in soc_link_dai_widgets. Improve the
consistency of the code by using snd_soc_dapm_connect_dai_link for both
types of DAIs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Wed, 5 Sep 2018 14:20:58 +0000 (15:20 +0100)]
ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked
Currently DAPM has a lot of similar code to handle errors from
snd_soc_dapm_new_control_unlocked, and much of this code does
not really accurately reflect what the function returns.
Firstly, most places will check for a return value of
-EPROBE_DEFER and silence any error messages in that case. The
one notable exception here being dapm_kcontrol_data_alloc
which does currently print any error messages in the case
of snd_soc_dapm_new_control_unlocked returning NULL or an
error. Additionally the error prints being silenced in these
case are redundant as snd_soc_dapm_new_control_unlocked can
only return -EPROBE_DEFER or NULL when failing.
Secondly, most places will treat a return value of NULL as
an -ENOMEM. This is not correct either since any error except
EPROBE_DEFER will cause a return value of NULL from
snd_soc_dapm_new_control_unlocked.
Centralise this handling and the error messages within
snd_soc_dapm_new_control_unlocked and update the callers
to simply check IS_ERR and return. Note that this update is
slightly simpler in the case of dapm_kcontrol_data_alloc where
that is fairly close to the handling that was already in place.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai.
To be more readable code, this patch adds
new for_each_rtd_codec_dai() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_dai_link has snd_soc_dai_link_component array
for codecs.
To be more readable code, this patch adds
new for_each_link_codecs() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Andrew F. Davis [Tue, 4 Sep 2018 15:36:17 +0000 (10:36 -0500)]
ASoC: tlv320aic31xx: Add overflow detection support
Similar to short circuit detection, when the ADC/DAC is saturated and
overflows poor audio quality can result and should be reported to the
user. This device support Automatic Dynamic Range Compression (DRC)
to reduce this but it is not enabled currently in this driver.
Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jon Hunter [Fri, 17 Aug 2018 15:35:43 +0000 (16:35 +0100)]
ASoC: core: Don't schedule DAPM work if already in target state
When dapm_power_widgets() is called, the dapm_pre_sequence_async() and
dapm_post_sequence_async() functions are scheduled for all DAPM contexts
(apart from the card DAPM context) regardless of whether the DAPM
context is already in the desired state. The overhead of this is not
insignificant and the more DAPM contexts there are the more overhead
there is.
For example, on the Tegra124 Jetson TK1, when profiling the time taken
to execute the dapm_power_widgets() the following times were observed.
Times for function dapm_power_widgets() are (us):
Min 23, Ave 190, Max 434, Count 39
Here 'Count' is the number of times that dapm_power_widgets() has been
called. Please note that the above time were measured using ktime_get()
to log the time on entry and exit from dapm_power_widgets(). So it
should be noted that these times may not be purely the time take to
execute this function if it is preempted. However, after applying this
patch and measuring the time taken to execute dapm_power_widgets() again
a significant improvement is seen as shown below.
Times for function dapm_power_widgets() are (us):
Min 4, Ave 16, Max 82, Count 39
Therefore, optimise the dapm_power_widgets() function by only scheduling
the dapm_pre/post_sequence_async() work if the DAPM context is not in
the desired state.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com> Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Main purpose of .nolock_start is we need to call
some function without spinlock.
OTOH we have .prepare which main purpose is
called under atomic context.
Then, it is called without spinlock.
In summary, our main callback init/quit, and start/stop
are called under "atomic context and with spinlock".
And some function need to be called under
"non-atomic context or without spinlock".
Let's merge .nolock_start and prepare to be more clear code.
Then, let's rename nolock_stop to cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:08:37 +0000 (07:08 +0000)]
ASoC: rsnd: ssiu: Support to init different BUSIF instance
Currently ssiu's .init is only called once during audio stream.
But SSIU with different BUSIF, shall be initialized each time,
even they are used in the same audio stream.
This patch introduces ssiu_status for BUSIF0 to BUSIF7 in rsnd_ssiu,
to make sure same .init for different BUSIF can always be executed.
To avoid the first stopped stream to stop the whole SSIU,
which may still has other BUSIF instance running, use usrcnt to count
the usage of SSIU, only the last user of SSIU can stop the whole SSIU.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:08:20 +0000 (07:08 +0000)]
ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0
Currently only BUSIF0 is supported by SSIU, all register setting
is done only for BUSIF.
Since BUSIF1 ~ BUSIF7 has been supported, so also support
these BUSIF from SSIU.
One note is that we can't support SSI9-4/5/6/7 so far,
because its address is out of calculation rule.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:08:00 +0000 (07:08 +0000)]
ASoc: rsnd: dma: Calculate PDMACHCRE with consider of BUSIF
PDMACHCR setting for SSI only considers BUSIF0 so far.
But BUSIF1 ~ BUSIF7 also maybe used, in the future.
This patch updates table gen2_id_table_ssiu, to also consider
BUSIF number used by SSI.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstreaming] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:07:43 +0000 (07:07 +0000)]
ASoc: rsnd: dma: Calculate dma address with consider of BUSIF
DMA address calculated by rsnd_dma_addr() only considers BUSIF0 so far.
But BUSIF1 ~ BUSIF7 also maybe used, in the future.
This patch updates DMA address calculations, to also consider
BUSIF number used by SSI.
One note is that we can't support SSI9-4/5/6/7 so far,
because its address is out of calculation rule.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstreaming] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:07:26 +0000 (07:07 +0000)]
ASoC: rsnd: ssi: Check runtime channel number rather than hw_params
The number of channel handled by SSI maybe differs from the one set
in hw_params, currently SSI checks hw_params's channel number,
and constrains to use same channel number, when it is being
used by multiple clients.
This patch corrects to check runtime channel number rather
than channel number set in hw_params.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstreaming] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:07:07 +0000 (07:07 +0000)]
ASoC: rsnd: ssi: Fix issue in dma data address assignment
Same SSI device may be used in different dai links,
by only having one dma struct in rsnd_ssi, after the first
instance's dma config be initilized, the following instances
can no longer configure dma, this causes issue, when their
dma data address are different from the first instance.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:06:50 +0000 (07:06 +0000)]
ASoC: rsnd: remove is_play parameter from hw_rule function
Currently rsnd_dai_stream *io is set to either &rdai->playback or
&rdai->capture based on whether it is a playback or capture stream,
in __rsnd_soc_hw_rule_* functions, but this is not necessary,
rsnd_dai_stream *io handler can be get from rule->private.
This patch removes 'is_play' parameter from hw_rule function.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:05:11 +0000 (07:05 +0000)]
ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime()
Add warning message to rsnd_kctrl_accept_runtime(), when kctrl
update is rejected due to corresponding dai-link is idle.
So that user can notice the reason of kctrl update failure.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Jiada Wang [Mon, 3 Sep 2018 07:08:58 +0000 (07:08 +0000)]
ASoC: rsnd: fixup not to call clk_get/set under non-atomic
Clocking operations clk_get/set_rate, are non-atomic,
they shouldn't be called in soc_pcm_trigger() which is atomic.
Following issue was found due to execution of clk_get_rate() causes
sleep in soc_pcm_trigger(), which shouldn't be blocked.
We can reproduce this issue by following
> enable CONFIG_DEBUG_ATOMIC_SLEEP=y
> compile, and boot
> mount -t debugfs none /sys/kernel/debug
> while true; do cat /sys/kernel/debug/clk/clk_summary > /dev/null; done &
> while true; do aplay xxx; done
This patch adds support to .prepare callback, and moves non-atomic
clocking operations to it. As .prepare is non-atomic, it is always
called before trigger_start/trigger_stop.
BUG: sleeping function called from invalid context at kernel/locking/mutex.c:620
in_atomic(): 1, irqs_disabled(): 128, pid: 2242, name: aplay
INFO: lockdep is turned off.
irq event stamp: 5964
hardirqs last enabled at (5963): [<ffff200008e59e40>] mutex_lock_nested+0x6e8/0x6f0
hardirqs last disabled at (5964): [<ffff200008e623f0>] _raw_spin_lock_irqsave+0x24/0x68
softirqs last enabled at (5502): [<ffff200008081838>] __do_softirq+0x560/0x10c0
softirqs last disabled at (5495): [<ffff2000080c2e78>] irq_exit+0x160/0x25c
Preemption disabled at:[ 62.904063] [<ffff200008be4d48>] snd_pcm_stream_lock+0xb4/0xc0
CPU: 2 PID: 2242 Comm: aplay Tainted: G B C 4.9.54+ #186
Hardware name: Renesas Salvator-X board based on r8a7795 (DT)
Call trace:
[<ffff20000808fe48>] dump_backtrace+0x0/0x37c
[<ffff2000080901d8>] show_stack+0x14/0x1c
[<ffff2000086f4458>] dump_stack+0xfc/0x154
[<ffff2000081134a0>] ___might_sleep+0x57c/0x58c
[<ffff2000081136b8>] __might_sleep+0x208/0x21c
[<ffff200008e5980c>] mutex_lock_nested+0xb4/0x6f0
[<ffff2000087cac74>] clk_prepare_lock+0xb0/0x184
[<ffff2000087cb094>] clk_core_get_rate+0x14/0x54
[<ffff2000087cb0f4>] clk_get_rate+0x20/0x34
[<ffff20000113aa00>] rsnd_adg_ssi_clk_try_start+0x158/0x4f8 [snd_soc_rcar]
[<ffff20000113da00>] rsnd_ssi_init+0x668/0x7a0 [snd_soc_rcar]
[<ffff200001133ff4>] rsnd_soc_dai_trigger+0x4bc/0xcf8 [snd_soc_rcar]
[<ffff200008c1af24>] soc_pcm_trigger+0x2a4/0x2d4
Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
Andrew F. Davis [Fri, 31 Aug 2018 18:24:31 +0000 (13:24 -0500)]
ASoC: tlv320aic31xx: Add short circuit detection support
These devices support detecting and reporting short circuits across
the output stages. Add support for reporting these issue. Do this
by registering an interrupt if available and enabling this error
to trigger that interrupt in the device.
Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the router data fields are left as default zeros which are
valid dai ids, so initialize these to invalid value of -1.
Without intializing these correctly get_session_from_id() can return
incorrect session resulting in not closing the opened copp and messing
up with the copp ref count.
Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Andrew F. Davis [Fri, 31 Aug 2018 15:14:05 +0000 (10:14 -0500)]
ASoC: tas6424: Save last fault register even when clear
When there is no fault bit set in a fault register we skip the fault
reporting section for that register. This also skips over saving that
registers value. We save the value so we will not double report an
error, but if an error clears then returns we will also not report it
as we did not save the all cleared register value. Fix this by saving
the fault register value in the all clear path.
Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
Andrew F. Davis [Fri, 31 Aug 2018 15:14:06 +0000 (10:14 -0500)]
ASoC: tas6424: Print full register name in error message
The current short version of the register name may be
ambiguous when another fault register detection is added.
Use the full name.
While here fix comment about clearing faults, the CLEAR_FAULT
register actually only clears sticky bits, which are only
warnings, fault bits can only cleared by resolving the fault.
Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-scu-card for platform.
ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-card for platform.
ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-scu-card for platform.
ASoC: simple-card: support snd_soc_dai_link_component style for platform
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card for platform.
ASoC: simple-card-util: support snd_soc_dai_link_component style for platform
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card-util for platform.
ASoC: soc-core: use snd_soc_dai_link_component for platform
Current struct snd_soc_dai_link is supporting multicodec,
and it is supporting legacy style of
codec_name
codec_of_node
code_dai_name
This is handled as single entry of multicodec.
We don't have multicpu support yet, but in the future we will.
In such case, we can use snd_soc_dai_link_component for both
cpu/codec. Then the code will be more simple and readble.
As next step, we want to use it for platform, too.
This patch adds snd_soc_dai_link_component style for platform.
We might have multiplatform support in the future, but we
don't know yet. To avoid un-known issue / complex code,
this patch supports just single-platform as 1st step.
If we could use snd_soc_dai_link_component for all CPU/Codec/Platform,
we will switch to new style, and remove legacy code.
This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for codec
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-scu-card for codec.
audi-graph-card and audio-graph-scu-card are very similar driver,
but using different feature. Thus we are keeping synchronization
on these 2 drivers style, because it is easy to confirm / check.
Current big difference between these 2 drivers are "dai_props" on
graph_card_data (= priv).
It will be difficult to keep synchronize if we will add new feature
on audio-graph-scu-card. Thus, this patch synchronize it.
ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-card for codec.
ASoC: simple-scu-card: support snd_soc_dai_link_component style for codec
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-scu-card for codec.
simple-card and simple-scu-card are very similar driver,
but using different feature. Thus we are keeping synchronization
on these 2 drivers style, because it is easy to confirm / check.
Current big difference between these 2 drivers are "dai_props" on
simple_card_data (= priv).
It will be difficult to keep synchronize if we will add new feature
on simple-scu-card. Thus, this patch synchronize it.
ASoC: simple-card: support snd_soc_dai_link_component style for codec
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card for codec.
ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple_card_utils for codec.
Matt Flax [Wed, 29 Aug 2018 23:38:02 +0000 (09:38 +1000)]
ASoC: cs4265: Add a S/PDIF enable switch
This patch adds a S/PDIF enable switch as a SOC_SINGLE.
Signed-off-by: Matt Flax <flatmax@flatmax.org> Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Matt Flax [Wed, 29 Aug 2018 23:38:01 +0000 (09:38 +1000)]
ASoC: cs4265: Add native 32bit I2S transport
The cs4265 uses 32 bit transport on the I2S bus. This patch enables native
32 bit mode for machine drivers which use this sound card driver.
Signed-off-by: Matt Flax <flatmax@flatmax.org> Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Matt Flax [Wed, 29 Aug 2018 23:38:00 +0000 (09:38 +1000)]
ASoC: cs4265: SOC_SINGLE register value error fix
The cs4265 driver declares the "MMTLR Data Switch" register setting with
a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2).
This incorrect value causes alsamixer to fault with the output :
cannot load mixer controls: Input/output error
This patch corrects the register value. alsamixer now runs.
Signed-off-by: Matt Flax <flatmax@flatmax.org> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: codecs: tas5720: add TAS5722 TDM slot width setting support
Unlike the TAS5720, the TAS5722 can be configured to utilize 16-bit wide
slots in TDM mode. This can help easing audio clocking/frequency
requirements.
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com> Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: codecs: tas5720: add TAS5722 specific volume control
The TAS5722 supports modifying volume in 0.25dB steps (as opposed to
0.5dB steps on the TAS5720). Introduce a custom mixer control that
allows taking advantage of this finer output volume granularity.
Also add custom getters/setters for access as the TAS5722 digital volume
controls are split over two registers.
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com> Signed-off-by: Andrew F. Davis <afd@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Akshu Agrawal [Tue, 21 Aug 2018 06:59:43 +0000 (12:29 +0530)]
ASoC: AMD: Change MCLK to 48Mhz
25Mhz MCLK which was earlier used was of spread type.
Thus, we were not getting accurate rate. The 48Mhz system
clk is of non-spread type and we are changing to it to get
accurate rate.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Reviewed-by: Daniel Kurtz <djkurtz@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds support for TI PCM3060 CODEC.
The technical documentation is available at [1].
[1] http://ti.com/product/pcm3060
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech> Cc: Mark Brown <broonie@kernel.org> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: M R Swami Reddy <mr.swami.reddy@ti.com> Cc: Vishwas A Deshpande <vishwas.a.deshpande@ti.com> Cc: Kevin Cernekee <cernekee@chromium.org> Cc: Peter Ujfalusi <peter.ujfalusi@ti.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
Fabrizio Castro [Tue, 21 Aug 2018 16:42:28 +0000 (17:42 +0100)]
ASoC: rsnd: Add r8a774a1 support
Document RZ/G2M (R8A774A1) SoC bindings.
Signed-off-by: Fabrizio Castro <fabrizio.castro@bp.renesas.com> Reviewed-by: Biju Das <biju.das@bp.renesas.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Simon Horman <horms+renesas@verge.net.au> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Tue, 21 Aug 2018 11:43:37 +0000 (13:43 +0200)]
ASoC: Intel: cht-bsw-rt5672: Add key-mappings for the headset buttons
Having the headset buttons send BTN_0, BTN_1 and BTN_2 events is not
really useful. Add mappings to PLAYPAUSE VOLUME_UP and VOLUME_DOWN like
we do in other Intel machine drivers.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Danny Smith [Tue, 21 Aug 2018 11:07:49 +0000 (13:07 +0200)]
ASoC: adau17x1: Implemented safeload support
Safeload support has been implemented which is used
when updating for instance filter parameters using
alsa controls. Without safeload support audio can
become distorted during update.
Signed-off-by: Danny Smith <dannys@axis.com> Signed-off-by: Robert Rosengren <robertr@axis.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Rakesh Ughreja [Wed, 22 Aug 2018 20:25:03 +0000 (15:25 -0500)]
ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers
This patch adds a kernel module which is used by the legacy HDA
codec drivers as library. This implements hdac_ext_bus_ops to enable
the reuse of legacy HDA codec drivers with ASoC platform drivers.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Rakesh Ughreja [Wed, 22 Aug 2018 20:25:00 +0000 (15:25 -0500)]
ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails
When no I2S based codec entries are found in the BIOS, check if there are
any HDA codecs detected on the bus. Based on the number of codecs found
take appropriate action in machine driver. If there are two HDA codecs
i.e. iDisp + HDA found on the bus, register DAIs and DAI links for both.
If only one codec i.e. iDisp is found then load only iDisp machine driver.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Rakesh Ughreja [Wed, 22 Aug 2018 20:24:59 +0000 (15:24 -0500)]
ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs
Add machine driver for Intel platforms (SKL/KBL/BXT/APL) with
HDA and iDisp codecs. This patch adds support for only iDisp (HDMI/DP)
codec. In the following patches support for HDA codecs will be added.
This should work for other Intel platforms as well e.g. GLK,CNL
however this series is not tested on all the platforms.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: common: add table for HDA-based platforms
Expose a table containing machine driver information for HDAudio-based
platforms handled by ASoC on Intel hardware.
We only set constant values that are valid across multiple
platforms. The firmware name used by the DSP will be set dynamically
for each platform.
The table is made of a single entry for now, if we need more
complicated set-up where HDAudio is mixed with ACPI-enumerated devices
(I2C, SoundWire) then we'd expect the differentiation to be handled
through information provided by the BIOS (as done for KBL
Chromebooks).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
HID made of either Wolfson/CirrusLogic PCI ID + 8804 identifier.
This helps enumerate the HifiBerry Digi+ HAT boards on the Up2 platform.
The scripts at https://github.com/thesofproject/acpi-scripts can be
used to add the ACPI initrd overlays.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 27 Aug 2018 13:26:47 +0000 (14:26 +0100)]
ASoC: dpcm: Properly initialise hw->rate_max
If the CPU DAI does not initialise rate_max, say if using
using KNOT or CONTINUOUS, then the rate_max field will be
initialised to 0. A value of zero in the rate_max field of
the hardware runtime will cause the sound card to support no
sample rates at all. Obviously this is not desired, just a
different mechanism is being used to apply the constraints. As
such update the setting of rate_max in dpcm_init_runtime_hw
to be consistent with the non-DPCM cases and set rate_max to
UINT_MAX if nothing is defined on the CPU DAI.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Tue, 28 Aug 2018 13:35:03 +0000 (14:35 +0100)]
ASoC: dapm: Don't fail creating new DAPM control on NULL pinctrl
devm_pinctrl_get will only return NULL in the case that pinctrl
is not built into the kernel and all the pinctrl functions used
by the DAPM core are appropriately stubbed for that case. There
is no need to error out of snd_soc_dapm_new_control_unlocked
if pinctrl isn't built into the kernel, so change the
IS_ERR_OR_NULL to just an IS_ERR.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
projects / mirror_ubuntu-kernels.git / log