Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 93 bytes.
Before:
text data bss dec hex filename
38961 9784 64 48809 bea9 sound/soc/codecs/rt1305.o
After:
text data bss dec hex filename
38804 9848 64 48716 be4c sound/soc/codecs/rt1305.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907074156.21907-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 100 bytes.
Before:
text data bss dec hex filename
51463 13016 128 64607 fc5f sound/soc/codecs/rt1011.o
After:
text data bss dec hex filename
51299 13080 128 64507 fbfb sound/soc/codecs/rt1011.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907073717.21632-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch set 0Hz to sysclk when shutdown the card.
Some codecs set rate constraints that derives from sysclk. This
mechanism works correctly if machine drivers give fixed frequency.
But simple-audio and audio-graph card set variable clock rate if
'mclk-fs' property exists. In this case, rate constraints will go
bad scenario. For example a codec accepts three limited rates
(mclk / 256, mclk / 384, mclk / 512).
Bad scenario as follows (mclk-fs = 256):
- Initialize sysclk by correct value (Ex. 12.288MHz)
- Codec set constraints of PCM rate by sysclk
48kHz (1/256), 32kHz (1/384), 24kHz (1/512)
- Play 48kHz sound, it's acceptable
- Sysclk is not changed
- Play 32kHz sound, it's acceptable
- Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256)
- Codec set constraints of PCM rate by sysclk
32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512)
- Play 48kHz again, but it's NOT acceptable because constraints
do not allow 48kHz
So codecs treat 0Hz sysclk as signal of applying no constraints to
avoid this problem.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907174501.19833-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch ignores sysclk setting if it is 0Hz.
Some codecs treat 0Hz sysclk as signal of applying no constraints.
This driver does not have such feature but current implementation
outputs 'Failed to set mclk' error message if machine driver sets
0Hz sysclk to this driver.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907174332.19586-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the arrays on the stack but instead make them
static const. Makes the object code smaller by 37 bytes.
Before:
text data bss dec hex filename
16253 7200 0 23453 5b9d sound/soc/codecs/ad193x.o
After:
text data bss dec hex filename
16056 7360 0 23416 5b78 sound/soc/codecs/ad193x.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190906161404.1440-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch supports some type of machine drivers that set 0 to mclk
when sound device goes to idle state. After applied this patch,
sysclk == 0 means there is no constraint of sound rate and other
values will set constraints which is derived by sysclk setting.
Original code refuses sysclk == 0 setting. But some boards and SoC
(such as RockPro64 and RockChip I2S) has connected SoC MCLK out to
ES8316 MCLK in. In this case, SoC side I2S will choose suitable
frequency of MCLK such as fs * mclk-fs when user starts playing or
capturing.
Bad scenario as follows (mclk-fs = 256):
- Initialize sysclk by correct value (Ex. 12.288MHz)
- ES8316 set constraints of PCM rate by sysclk
48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400),
24kHz (1/512), 16kHz (1/768), 12kHz (1/1024)
- Play 48kHz sound, it's acceptable
- Sysclk is not changed
- Play 32kHz sound, it's acceptable
- Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256)
- ES8316 set constraints of PCM rate by sysclk
32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400),
16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024)
- Play 48kHz again, but it's NOT acceptable because constraints
list does not allow 48kHz
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907163653.9382-2-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
When do compile test, if SND_SOC_SOF_OF is not set, we get:
sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_request':
imx8.c:(.text+0xb0): undefined reference to `snd_sof_ipc_msgs_rx'
sound/soc/sof/imx/imx8.o: In function `imx8_ipc_msg_data':
imx8.c:(.text+0xf4): undefined reference to `sof_mailbox_read'
sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_reply':
imx8.c:(.text+0x160): undefined reference to `sof_mailbox_read'
Make SND_SOC_SOF_IMX_TOPLEVEL always depends on SND_SOC_SOF_OF
Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 202acc565a ("ASoC: SOF: imx: Add i.MX8 HW support")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190905064400.24800-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The g12a audio subsystem, which is a derivative of the axg subsystem,
provides a dedicated reset line for each of the audio components.
The axg did not provide that and it is unclear if/when these reset are
required. The reset already helped solve a channel mapping issue on the
tdm formatter devices. Let's add the reset binding for the other
components, so we can describe this in DT. We'll use it later on
in the driver when/if needed.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform
were unbinded and re-binded again.
But, Audio Graph Card might can't rebind again if user do for example
unbind CPU or Codec driver
bind CPU or Codec driver
Because Audio Graph Card is still pointing old/unbinded
CPU or Codec driver's DAI name at dlc->dai_name.
To avoid this issue, it needs to alloc memory and keep DAI name
even though if CPU or Codec driver was unbinded.
Or, always do unbind/bind at Sound Card.
For now, this patch indicates this issue as FIXME.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgpdu75m.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform
were unbinded and re-binded again.
But, Simple Card might can't rebind again if user do for example
unbind CPU or Codec driver
bind CPU or Codec driver
Because Simple Card is still pointing old/unbinded
CPU or Codec driver's DAI name at dlc->dai_name.
To avoid this issue, it needs to alloc memory and keep DAI name
even though if CPU or Codec driver was unbinded.
Or, always do unbind/bind at Sound Card.
For now, this patch indicates this issue as FIXME.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tv9tu75x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
soc_probe_link_components() has paired soc_remove_link_components(),
but, these are implemented at different place.
So it is difficult to confirm code.
This patch moves soc_probe_link_components() next to
soc_remove_link_components().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o90g7lbd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
soc-dapm has snd_soc_dapm_free() which cleanups debugfs, widgets, list.
But, there is no paired initialize function.
This patch adds snd_soc_dapm_init() and initilaizing dapm
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnkw7lbj.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC setups some dapm related member at
snd_soc_component_initialize() which is called when component was
registered, and setups remaining member at soc_probe_component()
which is called when component was probed.
This kind of setup separation is no meanings, and it is very
difficult to read and confusable.
This patch setups all dapm settings at one place.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r25c7lbo.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
soc_probe_comonent() has paired soc_remove_comonent(),
but, these are implemented at different place.
So it is difficult to confirm code.
This patch moves soc_probe_component() next to
soc_remove_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgps7lbt.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
soc_rtd_init() was soc_post_component_init(), but there was no
its paired soc_post_component_free(), but it is done at
soc_remove_link_dais().
This means it is difficult to find related code.
This patch adds soc_rtd_free() which is paired soc_rtd_init().
soc_rtd_xxx() will be more cleanuped in the future.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tva87lby.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Allwinner A64 SoC has an embedded audio codec that uses a separate
controller to drive its analog part, which is supported in Linux, with a
matching Device Tree binding.
Now that we have the DT validation in place, let's convert the device tree
bindings for that controller over to a YAML schemas.
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Link: https://lore.kernel.org/r/20190828125209.28173-5-mripard@kernel.org
Reviewed-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The forward declaration of mt8183_mt6358_ts3a227_max98357_headset_init
is for cyclic dependency between card, headset_dev, and headset_init.
It used to be:
- card depends on headset_dev
- headset_dev depends on headset_init
- headset_init depends on card
Commit a962a809e5 ("ASoC: mediatek: mt8183: make headset codec
optional") removed the cyclic dependency.
Thus, it is safe to remove the forward declaration.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20190830074240.195166-4-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We should only select SND_INTEL_NHLT when ACPI is defined. This was
done for the legacy HDAudio driver but not for DSP-enabled cases,
leading to compilation errors with randconfig.
Fix by aligning on the same solution.
For the Skylake driver this is overkill since there is a top-level
dependency on ACPI, but it doesn't hurt and it's better to have
consistency.
Fixes: 68b953aeb5 ('ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number')
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190829214213.11653-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Apart from Haswell machines, all other devices have their private data
set to snd_soc_acpi_mach instance.
Changes for HSW/ BDW boards introduced with series:
https://patchwork.kernel.org/cover/10782035/
added support for dai_link platform_name adjustments within card probe
routines. These take for granted private_data points to
snd_soc_acpi_mach whereas for Haswell, it's sst_pdata instead. Change
private context of platform_device - representing machine board - to
address this.
Fixes: e87055d732 ("ASoC: Intel: haswell: platform name fixup support")
Fixes: 7e40ddcf97 ("ASoC: Intel: bdw-rt5677: platform name fixup support")
Fixes: 2d067b2807 ("ASoC: Intel: broadwell: platform name fixup support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190822113616.22702-2-cezary.rojewski@intel.com
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the daVinci devices does not boot with DT. In this case the DMA
channel is looked up with dma_slave_map and for that the chan_names[]
must be configured.
Both McASP and ASP/McBSP uses "tx" and "rx" as channel names, so we can
just do this when the dev->of_node is not valid.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
