[ALSA] ASoC: Clarify API for bias configuration

Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.

Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.

Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

2 files changed, 26 insertions, 5 deletions

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index bf4cf0c..f8223fa 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -221,7 +221,8 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
 	int event);
-int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event);
+int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+	enum snd_soc_bias_level level);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 73accbc..bca9538 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -103,6 +103,24 @@
 	.private_value = (unsigned long)&xenum }
 
 /*
+ * Bias levels
+ *
+ * @ON:      Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ *           stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ *           in progress. NOTE: The transition time between STANDBY and ON
+ *           should be as fast as possible and no longer than 10ms.
+ * @OFF:     Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+	SND_SOC_BIAS_ON,
+	SND_SOC_BIAS_PREPARE,
+	SND_SOC_BIAS_STANDBY,
+	SND_SOC_BIAS_OFF,
+};
+
+/*
  * Digital Audio Interface (DAI) types
  */
 #define SND_SOC_DAI_AC97	0x1
@@ -356,7 +374,8 @@ struct snd_soc_codec {
 	struct mutex mutex;
 
 	/* callbacks */
-	int (*dapm_event)(struct snd_soc_codec *codec, int event);
+	int (*set_bias_level)(struct snd_soc_codec *,
+			      enum snd_soc_bias_level level);
 
 	/* runtime */
 	struct snd_card *card;
@@ -378,8 +397,8 @@ struct snd_soc_codec {
 	/* dapm */
 	struct list_head dapm_widgets;
 	struct list_head dapm_paths;
-	unsigned int dapm_state;
-	unsigned int suspend_dapm_state;
+	enum snd_soc_bias_level bias_level;
+	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
 
 	/* codec DAI's */
@@ -449,7 +468,8 @@ struct snd_soc_machine {
 	int (*resume_post)(struct platform_device *pdev);
 
 	/* callbacks */
-	int (*dapm_event)(struct snd_soc_machine *, int event);
+	int (*set_bias_level)(struct snd_soc_machine *,
+			      enum snd_soc_bias_level level);
 
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;