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, 5 insertions, 4 deletions

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 01d19e9f53f9..a15a3e974f0d 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1172,7 +1172,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
                 break;
 
         case SND_SOC_BIAS_STANDBY:
-                if (codec->bias_level == SND_SOC_BIAS_OFF) {
+                if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
                         /* Enable Audio PLL & Audio section */
                         data = AUDIO_PLL | AUDIO_SECTION_RESET
                                 | AUDIO_SECTION_ON;
@@ -1185,7 +1185,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
                 pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
                 break;
         }
-        codec->bias_level = level;
+        codec->dapm.bias_level = level;
         return 0;
 }
 
@@ -1346,6 +1346,7 @@ EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
 static int pm860x_probe(struct snd_soc_codec *codec)
 {
         struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
         int i, ret;
 
         pm860x->codec = codec;
@@ -1374,9 +1375,9 @@ static int pm860x_probe(struct snd_soc_codec *codec)
 
         snd_soc_add_controls(codec, pm860x_snd_controls,
                              ARRAY_SIZE(pm860x_snd_controls));
-        snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
+        snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets,
                                   ARRAY_SIZE(pm860x_dapm_widgets));
-        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+        snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
         return 0;
 
 out_codec: