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>
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 73d0edd..0a9bd68 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -49,31 +49,33 @@
 
 static void tosa_ext_control(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
 	/* set up jack connection */
 	switch (tosa_jack_func) {
 	case TOSA_HP:
-		snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
-		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case TOSA_MIC_INT:
-		snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_disable_pin(dapm, "Headset Jack");
 		break;
 	case TOSA_HEADSET:
-		snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
-		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
-		snd_soc_dapm_enable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+		snd_soc_dapm_enable_pin(dapm, "Headset Jack");
 		break;
 	}
 
 	if (tosa_spk_func == TOSA_SPK_ON)
-		snd_soc_dapm_enable_pin(codec, "Speaker");
+		snd_soc_dapm_enable_pin(dapm, "Speaker");
 	else
-		snd_soc_dapm_disable_pin(codec, "Speaker");
+		snd_soc_dapm_disable_pin(dapm, "Speaker");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 }
 
 static int tosa_startup(struct snd_pcm_substream *substream)
@@ -186,10 +188,11 @@
 static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int err;
 
-	snd_soc_dapm_nc_pin(codec, "OUT3");
-	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+	snd_soc_dapm_nc_pin(dapm, "OUT3");
+	snd_soc_dapm_nc_pin(dapm, "MONOOUT");
 
 	/* add tosa specific controls */
 	err = snd_soc_add_controls(codec, tosa_controls,
@@ -198,13 +201,13 @@
 		return err;
 
 	/* add tosa specific widgets */
-	snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
 				  ARRAY_SIZE(tosa_dapm_widgets));
 
 	/* set up tosa specific audio path audio_map */
-	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 	return 0;
 }