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, 12 insertions, 11 deletions

diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index fac61744f8c7..5a45067b43ba 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -832,7 +832,7 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
 		break;
 	}
-	codec->bias_level = level;
+	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -888,10 +888,10 @@ static int alc5623_resume(struct snd_soc_codec *codec)
 	alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge alc5623 caps */
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+	if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
 		alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-		codec->bias_level = SND_SOC_BIAS_ON;
-		alc5623_set_bias_level(codec, codec->bias_level);
+		codec->dapm.bias_level = SND_SOC_BIAS_ON;
+		alc5623_set_bias_level(codec, codec->dapm.bias_level);
 	}
 
 	return 0;
@@ -900,6 +900,7 @@ static int alc5623_resume(struct snd_soc_codec *codec)
 static int alc5623_probe(struct snd_soc_codec *codec)
 {
 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
@@ -943,24 +944,24 @@ static int alc5623_probe(struct snd_soc_codec *codec)
 	snd_soc_add_controls(codec, alc5623_snd_controls,
 			ARRAY_SIZE(alc5623_snd_controls));
 
-	snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets,
+	snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
 					ARRAY_SIZE(alc5623_dapm_widgets));
 
 	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
 
 	switch (alc5623->id) {
 	default:
 	case 0x21:
 	case 0x22:
-		snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets,
+		snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
 					ARRAY_SIZE(alc5623_dapm_amp_widgets));
-		snd_soc_dapm_add_routes(codec, intercon_amp_spk,
-						ARRAY_SIZE(intercon_amp_spk));
+		snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
+					ARRAY_SIZE(intercon_amp_spk));
 		break;
 	case 0x23:
-		snd_soc_dapm_add_routes(codec, intercon_spk,
-						ARRAY_SIZE(intercon_spk));
+		snd_soc_dapm_add_routes(dapm, intercon_spk,
+					ARRAY_SIZE(intercon_spk));
 		break;
 	}