treewide: Replace GPLv2 boilerplate/reference with SPDX - rule 152

Based on 1 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 as published by
the free software foundation either version 2 of the license or at
your option any later version

extracted by the scancode license scanner the SPDX license identifier

GPL-2.0-or-later

has been chosen to replace the boilerplate/reference in 3029 file(s).

Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Allison Randal <allison@lohutok.net>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190527070032.746973796@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

ASoC: multi-component - ASoC Multi-Component Support

This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)

struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)

struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)

struct snd_soc_device ---> deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>

Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>

TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>

Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>

MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>

i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>

J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>

CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>

Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

ASoC: CX20442: push down machine independent line discipline bits

This corrected patch adds machine independent line discipline code, prevoiusly
exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442
codec driver. The code can be used as a standalone line discipline, or as a
set of codec specific functions called from machine's line discipline
callbacks. Anyway, the line discipline itself must be registered by a machine
driver.

Applies on top of the followup to my initial driver version:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html

Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Add support for Conexant CX20442-11 voice modem codec

This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.

This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.

The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.

Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.

Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.

[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html

Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>