Daniel Mack [Wed, 17 Oct 2018 11:37:03 +0000 (13:37 +0200)]
ASoC: sta32x: Add support for XTI clock
The STA32x chips feature an XTI clock input that needs to be stable before
the reset signal is released. Therefore, the chip driver needs to get a
handle to the clock. Instead of relying on other parts of the system to
enable the clock, let the codec driver grab a handle itself.
In order to keep existing boards working, clock support is made optional.
Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com> Acked-by: Jon Hunter <jonathanh@nvidia.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: sunxi: allow the sun8i-codec driver to be built on ARM64
Allwinner A64 uses the same digital codec part as in A33, so we need
to build this driver on ARM64 as well.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls
The internal codec on A64 is split into 2 parts. The analog path controls
are routed through an embedded custom register bus accessed through
the PRCM block.
Add an ASoC component driver for it. This should be tied to the codec
audio card as an auxiliary device.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: dt-binding: Add bindings for Allwinner A64 codec's analog path controls
The internal codec on Allwinner A64 is split into 2 parts. The
analog path controls are routed through an embedded custom register
bus accessed through the PRCM block just as on A23/A33/H3.
Add a binding for this hardware.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: sun8i-codec-analog: split regmap code into separate driver
It will be reused by sun50i-codec-analog later.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: sun8i-codec: Don't hardcode BCLK / LRCK ratio
BCLK / LRCK ratio should be sample size * channels, but it was
hardcoded to 32 (0x1 is 32 as per A33 and A64 datasheets).
Calculate it basing on sample size and number of channels.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Marcus Cooper [Wed, 17 Oct 2018 07:38:05 +0000 (00:38 -0700)]
ASoC: sun4i-i2s: Add compatibility with A64 codec I2S
The I2S block used for the audio codec in the A64 differs from other 3
I2S modules in A64 and isn't compatible with H3. But it is very similar
to what is found in A10(sun4i). However, its TX FIFO is
located at a different address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com> Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Akshu Agrawal [Mon, 15 Oct 2018 06:54:44 +0000 (12:24 +0530)]
ASoC: AMD: Add SND_JACK_LINEOUT jack type
Some 3 pole connectors report impedance greater than threshold of
1000Ohm. Thus, da7219 reports them as LINEOUT.
Adding the SND_JACK_LINEOUT type so that we don't fail to detect
any 3 pole jack type.
Also, changing
SND_JACK_HEADPHONE | SND_JACK_MICROPHONE -> SND_JACK_HEADSET
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
RIGHT_J only can handle 16bit data bits.
Current driver just errored if user requests non RIGHT_J
+ 16bit combination. But it is not useful for user.
This patch adds HW constraint for it, and avoid
error on such situation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
commit fb2815f44a9e ("ASoC: rsnd: add support for 16/24 bit slot widths")
added TDM width check, and return error if it was not 16/24/32 bit.
But it is too strict. This patch uses 32bit same as default.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Wed, 10 Oct 2018 08:37:13 +0000 (10:37 +0200)]
ASoC: max98988: add I2C dependency
max98988 only builds with I2C support enabled, otherwise we get a build error:
sound/soc/codecs/max98088.c:1789:1: error: data definition has no type or storage class [-Werror]
module_i2c_driver(max98088_i2c_driver);
^~~~~~~~~~~~~~~~~
sound/soc/codecs/max98088.c:1789:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/max98088.c:1789:1: error: parameter names (without types) in function declaration [-Werror]
sound/soc/codecs/max98088.c:1780:26: error: 'max98088_i2c_driver' defined but not used [-Werror=unused-variable]
Fixes: 24ae67c58250 ("ASoC: max98988: make it selectable") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Reviewed-by: Marco Felsch <m.felsch@pengutronix.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Andreas Färber [Fri, 5 Oct 2018 07:58:11 +0000 (09:58 +0200)]
ASoC: max98088: Add master clock handling
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Cc: Tushar Behera <tushar.behera@linaro.org> Signed-off-by: Andreas Färber <afaerber@suse.de> Acked-by: Tushar Behera <trblinux@gmail.com> Reviewed-by: Javier Martinez Canillas <javier.martinez@collabora.co.uk>
[m.felsch@pengutronix.de: move mclk request to i2c_probe]
[m.felsch@pengutronix.de: make use of snd_soc_component_get_bias_level()] Signed-off-by: Marco Felsch <m.felsch@pengutronix.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Thu, 4 Oct 2018 18:30:06 +0000 (20:30 +0200)]
ASoC: topology: Use the standard fall-through annotations
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation.
gcc can't understand the mixed texts, unfortunately.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Thu, 4 Oct 2018 18:30:03 +0000 (20:30 +0200)]
ASoC: pcm186x: Use the standard fall-through annotation
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation.
Unfortunately gcc doesn't understand the mixed comment lines.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Thu, 4 Oct 2018 18:30:02 +0000 (20:30 +0200)]
ASoC: adau1761: Use the standard fall-through annotation
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation
at the right place. It has to be put right before the next label.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Daniel Mack [Wed, 3 Oct 2018 19:32:34 +0000 (21:32 +0200)]
ASoC: add fault detect recovery property to DT bindings
The driver already has support for setting the FDRB bit in the CONFA
register through platform data, but there was no property to set it
in the device-tree bindings.
Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Dan Carpenter [Mon, 1 Oct 2018 16:44:30 +0000 (19:44 +0300)]
ASoC: qdsp6: q6asm-dai: checking NULL vs IS_ERR()
The q6asm_audio_client_alloc() doesn't return NULL, it returns error
pointers.
Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Biju Das [Thu, 27 Sep 2018 13:51:25 +0000 (14:51 +0100)]
ASoC: rsnd: Add r8a7744 support
Document RZ/G1N (R8A7744) SoC bindings.
Signed-off-by: Biju Das <biju.das@bp.renesas.com> Reviewed-by: Chris Paterson <Chris.Paterson2@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
The 'ret' variable is now only used in an #ifdef, and causes a
warning if it is declared outside of that block:
sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe':
sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable]
Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes: a45f8853a5f9 ("ASoC: Add driver for PROTO Audio CODEC (with a WM8731)") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Signed-off-by: Mark Brown <broonie@kernel.org>
I don't know if that combination is supposed to work.
Assuming it is not, this adds a dependency on all users
for PXA to avoids the combination.
Fixes: 1c8bc7b3de5e ("ASoC: pxa: switch to new ac97 bus support") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: soc-utils: Rename dummy_dma_ops to snd_dummy_dma_ops
The symbols 'dummy_dma_ops' is declared with different data types by
sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This
leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h:
sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops'
static const struct snd_pcm_ops dummy_dma_ops = {
^
...
arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here
extern const struct dma_map_ops dummy_dma_ops;
^
Rename the symbol in soc-utils.c to 'snd_dummy_dma_ops' to avoid the
conflict.
Signed-off-by: Matthias Kaehlcke <mka@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Andreas Färber [Tue, 25 Sep 2018 14:23:48 +0000 (16:23 +0200)]
ASoC: dt-bindings: add max98088 audio codec
This patch adds the bindings for maxim max98088/9 audio codec.
Signed-off-by: Andreas Färber <afaerber@suse.de>
[m.felsch@pengutronix.de: adapt commit message]
[m.felsch@pengutronix.de: adapt formatting] Signed-off-by: Marco Felsch <m.felsch@pengutronix.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Matt Flax [Tue, 25 Sep 2018 06:40:18 +0000 (16:40 +1000)]
ASoC: cs4265: Add a MIC pre. route
The cs4265 driver is missing a microphone preamp enable.
This patch enables/disables the microphone preamp when mic
selection is made using the kcontrol.
Signed-off-by: Matt Flax <flatmax@flatmax.org> Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: rsnd: fixup SSI clock during suspend/resume modes
Prepare <-> Cleanup functions pair has balanced calls.
But in case of suspend mode no call to rsnd_soc_dai_shutdown()
function, so cleanup isn't called. OTOH during resume mode
function rsnd_soc_dai_prepare() is called, but calling
rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero,
bacause was not cleanup before).
We need to call rsnd_ssi_prepare(), because it enables SSI clocks
by calling rsnd_ssi_master_clk_start().
This patch allows to call prepare/cleanup functions always.
Signed-off-by: Dmytro Prokopchuk <dmytro.prokopchuk@globallogic.com> Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
[kuninori: adjusted to upstream] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: rename for_each_rtd_codec_dai_reverse to rollback
commit 0b7990e38971 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai_reverse(). but _rollback() is better
naming than _reverse(). This patch rename it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: convert for_each_rtd_codec_dai() for missing part
commit 0b7990e38971 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai(), but it didn't convert few loop
which is not using "rtd". This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: q6afe: dt-bindings: Update input range for qcom,sd-lines
Input to qcom,sd-lines should be between 0 and 3 instead of
1 to 4 as 0 corresponds to BIT(0) which is MI2S_SD0 line.
Bit 1 to 3 corresponds to SD1 to SD3 lines respectively.
Updated documentation for the same.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Shuming Fan [Tue, 18 Sep 2018 11:51:24 +0000 (19:51 +0800)]
ASoC: rt5682: Remove HP volume control
This patch removed Headphone Playback Volume control.
Due to codec settings, we don't want the user to change HP analog gain.
The user could use DAC1 Playback Volume control to
change playback volume.
Signed-off-by: Shuming Fan <shumingf@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
module.h already contained moduleparam.h, so it is safe to remove
the redundant include.
The issue is detected with the help of Coccinelle.
Signed-off-by: zhong jiang <zhongjiang@huawei.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Oder Chiou [Mon, 17 Sep 2018 11:03:09 +0000 (19:03 +0800)]
ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly
The value of period_bytes will get the zero before the hw_params() is not
run completely. Move the function snd_pcm_lib_period_bytes() to copy work,
and make sure that is not zero.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
To find (CPU/)Codec/Platform, we need to find component first
(= on CPU/Codec/Platform), and find DAI from it (= CPU/Codec).
These are similar operation but difficult to be simple,
and has many duplicate code to finding component.
This patch adds new snd_soc_is_matching_component(),
and reduce duplicate codes.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: soc-core: manage platform name under snd_soc_init_platform()
Now "platform" is controlled by snd_soc_dai_link_component,
thus its "name" can be initialized in snd_soc_init_platform(),
instead of soc_bind_dai_link() local.
This patch do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Wed, 12 Sep 2018 11:31:32 +0000 (12:31 +0100)]
ALSA: hda: Fix implicit definition of pci_iomap() on SH
Include asm/io.h directly so we've got a definition of pci_iomap(), the
current set of includes do this implicitly on most architectures but not
on SH.
Reported-by: kbuild test robot <lkp@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
sound: don't call skl_init_chip() to reset intel skl soc
Internally, skl_init_chip() calls snd_hdac_bus_init_chip() which
1) sets bus->chip_init to prevent multiple entrances before device
is stopped; 2) enables interrupt.
We shouldn't use it for the purpose of resetting device only because
1) when we really want to initialize device, we won't be able to do
so; 2) we are ready to handle interrupt yet, and kernel crashes when
interrupt comes in.
Rename azx_reset() to snd_hdac_bus_reset_link(), and use it to reset
device properly.
Fixes: 60767abcea3d ("ASoC: Intel: Skylake: Reset the controller in probe") Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Yu Zhao <yuzhao@google.com> Signed-off-by: Mark Brown <broonie@kernel.org>
sound: enable interrupt after dma buffer initialization
In snd_hdac_bus_init_chip(), we enable interrupt before
snd_hdac_bus_init_cmd_io() initializing dma buffers. If irq has
been acquired and irq handler uses the dma buffer, kernel may crash
when interrupt comes in.
Fix the problem by postponing enabling irq after dma buffer
initialization. And warn once on null dma buffer pointer during the
initialization.
Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Yu Zhao <yuzhao@google.com> Signed-off-by: Mark Brown <broonie@kernel.org>
The patch doesn't fix accessing memory with null pointer in
skl_interrupt().
There are two problems: 1) skl_init_chip() is called twice, before
and after dma buffer is allocate. The first call sets bus->chip_init
which prevents the second from initializing bus->corb.buf and
rirb.buf from bus->rb.area. 2) snd_hdac_bus_init_chip() enables
interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers.
There is a small window which skl_interrupt() can be called if irq
has been acquired. If so, it crashes when using null dma buffer
pointers.
Will fix the problems in the following patches. Also attaching the
crash for future reference.
It is strange if it has "dai" but doesn't have "dai->driver".
And more over "dai->driver->xxx" is used everywhere without
"dai->driver" pointer NULL checking.
It got Oops already if "dai->driver" was NULL.
Let's remove un-needed "dai->driver" NULL check.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Current behaviour of ASoC core w.r.t to component removal is that it
unregisters dependent sound card totally. There is no support to
rebind the card if the component comes back.
Typical use case is DSP restart or kernel modules itself.
With this patch, core now maintains list of cards that are unbind due to
any of its depended components are removed and card not unregistered yet.
This list is cleared when the card is rebind successfully or when the
card is unregistered from machine driver.
This list of unbind cards are tried to bind once again after every new
component is successfully added, giving a fair chance for card bind
to be successful.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Fix UBSAN warning at snd_soc_get/put_volsw_sx()
In functions snd_soc_get_volsw_sx() or snd_soc_put_volsw_sx(),
if the result of (min + max) is negative, then fls() returns
signed integer with value as 32. This leads to signed integer
overflow as complete operation is considered as signed integer.
UBSAN: Undefined behaviour in sound/soc/soc-ops.c:382:50
signed integer overflow:
-2147483648 - 1 cannot be represented in type 'int'
Call trace:
[<ffffff852f746fe4>] __dump_stack lib/dump_stack.c:15 [inline]
[<ffffff852f746fe4>] dump_stack+0xec/0x158 lib/dump_stack.c:51
[<ffffff852f7b5f3c>] ubsan_epilogue+0x18/0x50 lib/ubsan.c:164
[<ffffff852f7b6840>] handle_overflow+0xf8/0x130 lib/ubsan.c:195
[<ffffff852f7b68f0>] __ubsan_handle_sub_overflow+0x34/0x44 lib/ubsan.c:211
[<ffffff85307971a0>] snd_soc_get_volsw_sx+0x1a8/0x1f8 sound/soc/soc-ops.c:382
Typecast the operation to unsigned int to fix the issue.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: AMD: Fix simultaneous playback and capture on different channel
If capture and playback are started on different channel (I2S/BT)
there is a possibilty that channel information passed from machine driver
is overwritten before the configuration is done in dma driver.
Example:
113.597588: cz_max_startup: ---playback sets BT channel
113.597694: cz_dmic1_startup: ---capture sets I2S channel
113.597979: acp_dma_hw_params: ---configures capture for I2S channel
113.598114: acp_dma_hw_params: ---configures playback for I2S channel
This is fixed by having 2 separate instance for playback and capture.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Mon, 10 Sep 2018 14:28:39 +0000 (15:28 +0100)]
ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai
snd_soc_dapm_new_dai may return an error pointer and currently this
isn't checked for in dapm_connect_dai_link_widgets. Add code to check
the return value and not add routes in that case.
Fixes: 778ff5bb8689 ("ASoC: dapm: Move connection of CODEC to CODEC DAIs") Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
projects / mirror_ubuntu-kernels.git / log