<feed xmlns='http://www.w3.org/2005/Atom'>
<title>linux-dev/sound/soc/codecs/es8328.c, branch linus/master</title>
<subtitle>Linux kernel development work - see feature branches</subtitle>
<id>https://git.zx2c4.com/linux-dev/atom/sound/soc/codecs/es8328.c?h=linus%2Fmaster</id>
<link rel='self' href='https://git.zx2c4.com/linux-dev/atom/sound/soc/codecs/es8328.c?h=linus%2Fmaster'/>
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<updated>2022-06-06T11:31:27Z</updated>
<entry>
<title>ASoC: es8328: Fix event generation for deemphasis control</title>
<updated>2022-06-06T11:31:27Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@kernel.org</email>
</author>
<published>2022-06-03T12:39:37Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=8259610c2ec01c5cbfb61882ae176aabacac9c19'/>
<id>urn:sha1:8259610c2ec01c5cbfb61882ae176aabacac9c19</id>
<content type='text'>
Currently the put() method for the deemphasis control returns 0 when a new
value is written to the control even if the value changed, meaning events
are not generated. Fix this, skip the work of updating the value when it is
unchanged and then return 1 after having done so.

Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: es8328: Use modern ASoC DAI format terminology</title>
<updated>2022-02-28T13:33:46Z</updated>
<author>
<name>Mark Brown</name>
<email>broonie@kernel.org</email>
</author>
<published>2022-02-22T22:35:34Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=6d2608804ebe3516f9dd30c5428b11d9bc121764'/>
<id>urn:sha1:6d2608804ebe3516f9dd30c5428b11d9bc121764</id>
<content type='text'>
As part of moving to remove the old style defines for the bus clocks update
the es8328 driver to use more modern terminology for clocking.

Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
Link: https://lore.kernel.org/r/20220222223534.3212743-1-broonie@kernel.org
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: es*: sync parameter naming (rate/sample_bits)</title>
<updated>2021-01-21T12:38:02Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2021-01-15T04:54:39Z</published>
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<id>urn:sha1:4060064e8baf047a64b6286f5fd87cb0a34d8b89</id>
<content type='text'>
This patch syncs naming rule.

 - xxx_rates;
 + xxx_rate;

 - xxx_samplebits;
 + xxx_sample_bits;

Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Link: https://lore.kernel.org/r/874kjiolga.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: es8328: Remove redundant null check before clk_disable_unprepare</title>
<updated>2021-01-08T14:00:43Z</updated>
<author>
<name>Xu Wang</name>
<email>vulab@iscas.ac.cn</email>
</author>
<published>2021-01-08T08:58:34Z</published>
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<id>urn:sha1:31c51a424f5163ee6f14fcc251f81078457123e1</id>
<content type='text'>
Because clk_disable_unprepare() already checked NULL clock parameter,
so the additional check is unnecessary, just remove it.

Signed-off-by: Xu Wang &lt;vulab@iscas.ac.cn&gt;
Link: https://lore.kernel.org/r/20210108085834.7168-1-vulab@iscas.ac.cn
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: codecs: es*: merge .digital_mute() into .mute_stream()</title>
<updated>2020-07-16T22:06:17Z</updated>
<author>
<name>Kuninori Morimoto</name>
<email>kuninori.morimoto.gx@renesas.com</email>
</author>
<published>2020-07-09T01:56:57Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=8667d948a6185fcde88ebc89e090fa574a44c366'/>
<id>urn:sha1:8667d948a6185fcde88ebc89e090fa574a44c366</id>
<content type='text'>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream

	int snd_soc_dai_digital_mute(xxx, int direction)
	{
		...
		else if (dai-&gt;driver-&gt;ops-&gt;mute_stream)
(1)			return dai-&gt;driver-&gt;ops-&gt;mute_stream(xxx, direction);
		else if (direction == SNDRV_PCM_STREAM_PLAYBACK &amp;&amp;
			 dai-&gt;driver-&gt;ops-&gt;digital_mute)
(2)			return dai-&gt;driver-&gt;ops-&gt;digital_mute(xxx);
		...
	}

Signed-off-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Reviewed-by: Peter Ujfalusi &lt;peter.ujfalusi@ti.com&gt;
Link: https://lore.kernel.org/r/87tuyhwiwm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: codecs: Use fallthrough pseudo-keyword</title>
<updated>2020-07-09T21:20:31Z</updated>
<author>
<name>Gustavo A. R. Silva</name>
<email>gustavoars@kernel.org</email>
</author>
<published>2020-07-09T01:03:59Z</published>
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<id>urn:sha1:3e146b55a4f5213b5da0f243813efb380fa7f84d</id>
<content type='text'>
Replace the existing /* fall through */ comments and its variants with
the new pseudo-keyword macro fallthrough[1].

[1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fallthrough#implicit-switch-case-fall-through

Signed-off-by: Gustavo A. R. Silva &lt;gustavoars@kernel.org&gt;
Link: https://lore.kernel.org/r/20200709010359.GA18971@embeddedor
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: es8328: Fix copy-paste error in es8328_right_line_controls</title>
<updated>2019-08-15T14:06:49Z</updated>
<author>
<name>YueHaibing</name>
<email>yuehaibing@huawei.com</email>
</author>
<published>2019-08-15T09:23:00Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=d63887bc4f50fede7013bda7c733d58ecc43efc1'/>
<id>urn:sha1:d63887bc4f50fede7013bda7c733d58ecc43efc1</id>
<content type='text'>
It seems 'es8328_rline_enum' should be used
in es8328_right_line_controls

Fixes: 567e4f98922c ("ASoC: add es8328 codec driver")
Signed-off-by: YueHaibing &lt;yuehaibing@huawei.com&gt;
Link: https://lore.kernel.org/r/20190815092300.68712-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: es8328: remove unused variable 'pga_tlv'</title>
<updated>2019-08-15T14:06:31Z</updated>
<author>
<name>YueHaibing</name>
<email>yuehaibing@huawei.com</email>
</author>
<published>2019-08-15T09:20:56Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=97d39be9ce5befc2e36f15d0df33832e0f633565'/>
<id>urn:sha1:97d39be9ce5befc2e36f15d0df33832e0f633565</id>
<content type='text'>
sound/soc/codecs/es8328.c:102:35: warning:
 pga_tlv defined but not used [-Wunused-const-variable=]

They are never used, so can be removed.

Reported-by: Hulk Robot &lt;hulkci@huawei.com&gt;
Signed-off-by: YueHaibing &lt;yuehaibing@huawei.com&gt;
Link: https://lore.kernel.org/r/20190815092056.28724-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 500</title>
<updated>2019-06-19T15:09:55Z</updated>
<author>
<name>Thomas Gleixner</name>
<email>tglx@linutronix.de</email>
</author>
<published>2019-06-04T08:11:33Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=d2912cb15bdda8ba4a5dd73396ad62641af2f520'/>
<id>urn:sha1:d2912cb15bdda8ba4a5dd73396ad62641af2f520</id>
<content type='text'>
Based on 2 normalized pattern(s):

  this program is free software you can redistribute it and or modify
  it under the terms of the gnu general public license version 2 as
  published by the free software foundation

  this program is free software you can redistribute it and or modify
  it under the terms of the gnu general public license version 2 as
  published by the free software foundation #

extracted by the scancode license scanner the SPDX license identifier

  GPL-2.0-only

has been chosen to replace the boilerplate/reference in 4122 file(s).

Signed-off-by: Thomas Gleixner &lt;tglx@linutronix.de&gt;
Reviewed-by: Enrico Weigelt &lt;info@metux.net&gt;
Reviewed-by: Kate Stewart &lt;kstewart@linuxfoundation.org&gt;
Reviewed-by: Allison Randal &lt;allison@lohutok.net&gt;
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190604081206.933168790@linutronix.de
Signed-off-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
</content>
</entry>
<entry>
<title>Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound</title>
<updated>2018-10-25T16:00:15Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2018-10-25T16:00:15Z</published>
<link rel='alternate' type='text/html' href='https://git.zx2c4.com/linux-dev/commit/?id=3acbd2de6bc3af215c6ed7732dfc097d1e238503'/>
<id>urn:sha1:3acbd2de6bc3af215c6ed7732dfc097d1e238503</id>
<content type='text'>
Pull sound updates from Takashi Iwai:
 "There have been little changes in ALSA core stuff, but ASoC core still
  kept rolling for the continued restructuring. The rest are lots of
  small driver-specific changes and some minor API updates. Here are
  highlights:

  General:
  - Appropriate fall-through annotations everywhere
  - Some code cleanup in memalloc code, handling non-cacahed pages more
    commonly in the helper
  - Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently

  Drivers:
  - More HD-audio CA0132 codec improvement for supporting other Creative
    boards
  - Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
    give move support of existing HD-audio devices with DSP
  - A few device-specific HD-audio quirks as usual
  - New quirk for RME CC devices and correction for B&amp;W PX for USB-audio
  - FireWire: code refactoring including devres usages

  ASoC Core:
  - Continued componentization works; it's almost done!
  - A bunch of new for_each_foo macros
  - Cleanups and fixes in DAPM code

  ASoC Drivers:
  - MCLK support for several different devices, including CS42L51, STM32
    SAI, and MAX98373
  - Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
    MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
    TI PCM3060"

* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
  ASoC: stm32: sai: fix master clock naming
  ASoC: stm32: add clock dependency for sai
  ALSA: hda/ca0132 - Actually fix microphone issue
  ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
  ASoC: wm2000: Remove wm2000_read helper function
  ASoC: cs42l51: fix mclk support
  ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
  ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
  ASoC: cs42l51: add mclk support
  ASoC: stm32: sai: set sai as mclk clock provider
  ASoC: dt-bindings: add mclk support to cs42l51
  ASoC: dt-bindings: add mclk provider support to stm32 sai
  ASoC: soc-core: fix trivial checkpatch issues
  ASoC: dapm: Add support for hw_free on CODEC to CODEC links
  ASoC: Intel: kbl_da7219_max98927: minor white space clean up
  ALSA: i2c/cs8427: Fix int to char conversion
  ALSA: doc: Brush up the old writing-an-alsa-driver
  ASoC: rsnd: tidyup SSICR::SWSP for TDM
  ASoC: rsnd: enable TDM settings for SSI parent
  ASoC: pcm3168a: add hw constraint for capture channel
  ...
</content>
</entry>
</feed>
