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>

3 files changed, 22 insertions, 19 deletions

diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 5f4e59f4461..aede7e74ec3 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -318,27 +318,28 @@ static const struct snd_soc_dapm_route intercon[] = {
 static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int i;
 
 	/*
 	 * Add DAPM widgets
 	 */
 	for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]);
+		snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
 
 
 
 	/*
 	 * Setup audio path interconnects
 	 */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 
 
 	/* always connected pins */
-	snd_soc_dapm_enable_pin(codec, "Int Mic");
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_enable_pin(dapm, "Int Mic");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_sync(dapm);
 
 
 
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569dfd0e..da9c3037496 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -140,6 +140,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	printk(KERN_DEBUG
@@ -154,25 +155,25 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 	}
 
 	/* Add specific widgets */
-	snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets,
 				  ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
 	/* Set up specific audio path interconnects */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	/* not connected */
-	snd_soc_dapm_nc_pin(codec, "RLINEIN");
-	snd_soc_dapm_nc_pin(codec, "LLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+	snd_soc_dapm_nc_pin(dapm, "LLINEIN");
 
 #ifdef ENABLE_MIC_INPUT
-	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(dapm, "Int Mic");
 #else
-	snd_soc_dapm_nc_pin(codec, "Int Mic");
+	snd_soc_dapm_nc_pin(dapm, "Int Mic");
 #endif
 
 	/* always connected */
-	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d283561c1..92c709ed096 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -105,19 +105,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
 static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Add afeb9260 specific widgets */
-	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
 				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
 
 	/* Set up afeb9260 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_enable_pin(codec, "Headphone Jack");
-	snd_soc_dapm_enable_pin(codec, "Line In");
-	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	snd_soc_dapm_enable_pin(dapm, "Line In");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
-	snd_soc_dapm_sync(codec);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }