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authorLiam Girdwood <lrg@slimlogic.co.uk>2010-11-05 16:53:46 +0300
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-11-06 18:28:29 +0300
commitce6120cca2589ede530200c7cfe11ac9f144333c (patch)
tree6ea7c26ce64dd4753e7cf9a3b048e74614b169dc /sound/soc/codecs/wm8400.c
parent22e2fda5660cdf62513acabdb5c82a5af415f838 (diff)
downloadlinux-ce6120cca2589ede530200c7cfe11ac9f144333c.tar.xz
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>
Diffstat (limited to 'sound/soc/codecs/wm8400.c')
-rw-r--r--sound/soc/codecs/wm8400.c11
1 files changed, 6 insertions, 5 deletions
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 850299786e02..96927a457a34 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -911,10 +911,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int wm8400_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
- ARRAY_SIZE(wm8400_dapm_widgets));
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_controls(dapm, wm8400_dapm_widgets,
+ ARRAY_SIZE(wm8400_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
@@ -1219,7 +1220,7 @@ static int wm8400_set_bias_level(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) {
ret = regulator_bulk_enable(ARRAY_SIZE(power),
&power[0]);
if (ret != 0) {
@@ -1306,7 +1307,7 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->bias_level = level;
+ codec->dapm.bias_level = level;
return 0;
}