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/magician.c b/sound/soc/pxa/magician.c
index b8207ce..f1acdc5 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -44,27 +44,29 @@
static void magician_ext_control(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
if (magician_spk_switch)
- 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");
if (magician_hp_switch)
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
- snd_soc_dapm_disable_pin(codec, "Headset Mic");
- snd_soc_dapm_enable_pin(codec, "Call Mic");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
- snd_soc_dapm_disable_pin(codec, "Call Mic");
- snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
break;
}
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
}
static int magician_startup(struct snd_pcm_substream *substream)
@@ -395,15 +397,16 @@
static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* NC codec pins */
- snd_soc_dapm_nc_pin(codec, "VOUTLHP");
- snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+ snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
+ snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
/* FIXME: is anything connected here? */
- snd_soc_dapm_nc_pin(codec, "VINL");
- snd_soc_dapm_nc_pin(codec, "VINR");
+ snd_soc_dapm_nc_pin(dapm, "VINL");
+ snd_soc_dapm_nc_pin(dapm, "VINR");
/* Add magician specific controls */
err = snd_soc_add_controls(codec, uda1380_magician_controls,
@@ -412,13 +415,13 @@
return err;
/* Add magician specific widgets */
- snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
/* Set up magician specific audio path interconnects */
- 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;
}