ASoC: Decouple DAPM from CODECs

Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

1 files changed, 6 insertions, 5 deletions

diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 4f326f604104..79b650945bb2 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -882,7 +882,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
                 break;
 
         case SND_SOC_BIAS_PREPARE:
-                if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
+                if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
                         /* Enable bias generation */
                         reg = snd_soc_read(codec, WM8961_ANTI_POP);
                         reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
@@ -897,7 +897,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
                 break;
 
         case SND_SOC_BIAS_STANDBY:
-                if (codec->bias_level == SND_SOC_BIAS_PREPARE) {
+                if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
                         /* VREF off */
                         reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
                         reg &= ~WM8961_VREF;
@@ -919,7 +919,7 @@ static int wm8961_set_bias_level(struct snd_soc_codec *codec,
                 break;
         }
 
-        codec->bias_level = level;
+        codec->dapm.bias_level = level;
 
         return 0;
 }
@@ -959,6 +959,7 @@ static struct snd_soc_dai_driver wm8961_dai = {
 
 static int wm8961_probe(struct snd_soc_codec *codec)
 {
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
         int ret = 0;
         u16 reg;
 
@@ -1024,9 +1025,9 @@ static int wm8961_probe(struct snd_soc_codec *codec)
 
         snd_soc_add_controls(codec, wm8961_snd_controls,
                                 ARRAY_SIZE(wm8961_snd_controls));
-        snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets,
+        snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets,
                                   ARRAY_SIZE(wm8961_dapm_widgets));
-        snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+        snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
 
         return 0;
 }