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, 13 insertions, 12 deletions

diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8d5efb333c33..0ea578815003 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -388,27 +388,28 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 {
         struct wm8960_data *pdata = codec->dev->platform_data;
         struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
         struct snd_soc_dapm_widget *w;
 
-        snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+        snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets,
                                   ARRAY_SIZE(wm8960_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));
 
         /* In capless mode OUT3 is used to provide VMID for the
          * headphone outputs, otherwise it is used as a mono mixer.
          */
         if (pdata && pdata->capless) {
-                snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_capless,
+                snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_capless,
                                           ARRAY_SIZE(wm8960_dapm_widgets_capless));
 
-                snd_soc_dapm_add_routes(codec, audio_paths_capless,
+                snd_soc_dapm_add_routes(dapm, audio_paths_capless,
                                         ARRAY_SIZE(audio_paths_capless));
         } else {
-                snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_out3,
+                snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_out3,
                                           ARRAY_SIZE(wm8960_dapm_widgets_out3));
 
-                snd_soc_dapm_add_routes(codec, audio_paths_out3,
+                snd_soc_dapm_add_routes(dapm, audio_paths_out3,
                                         ARRAY_SIZE(audio_paths_out3));
         }
 
@@ -417,7 +418,7 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
          * list each time to find the desired power state do so now
          * and save the result.
          */
-        list_for_each_entry(w, &codec->dapm_widgets, list) {
+        list_for_each_entry(w, &codec->dapm.widgets, list) {
                 if (strcmp(w->name, "LOUT1 PGA") == 0)
                         wm8960->lout1 = w;
                 if (strcmp(w->name, "ROUT1 PGA") == 0)
@@ -572,7 +573,7 @@ static int wm8960_set_bias_level_out3(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 anti-pop features */
                         snd_soc_write(codec, WM8960_APOP1,
                                       WM8960_POBCTRL | WM8960_SOFT_ST |
@@ -610,7 +611,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
                 break;
         }
 
-        codec->bias_level = level;
+        codec->dapm.bias_level = level;
 
         return 0;
 }
@@ -626,7 +627,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
                 break;
 
         case SND_SOC_BIAS_PREPARE:
-                switch (codec->bias_level) {
+                switch (codec->dapm.bias_level) {
                 case SND_SOC_BIAS_STANDBY:
                         /* Enable anti pop mode */
                         snd_soc_update_bits(codec, WM8960_APOP1,
@@ -681,7 +682,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
                 break;
 
         case SND_SOC_BIAS_STANDBY:
-                switch (codec->bias_level) {
+                switch (codec->dapm.bias_level) {
                 case SND_SOC_BIAS_PREPARE:
                         /* Disable HP discharge */
                         snd_soc_update_bits(codec, WM8960_APOP2,
@@ -705,7 +706,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
                 break;
         }
 
-        codec->bias_level = level;
+        codec->dapm.bias_level = level;
 
         return 0;
 }