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authorLiam Girdwood <lrg@slimlogic.co.uk>2010-11-05 09:53:46 -0400
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-11-06 11:28:29 -0400
commitce6120cca2589ede530200c7cfe11ac9f144333c (patch)
tree6ea7c26ce64dd4753e7cf9a3b048e74614b169dc /sound/soc/codecs/ak4671.c
parent22e2fda5660cdf62513acabdb5c82a5af415f838 (diff)
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/ak4671.c')
-rw-r--r--sound/soc/codecs/ak4671.c9
1 files changed, 5 insertions, 4 deletions
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 24f5f49bb9d..1d6573c38af 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -437,10 +437,11 @@ static const struct snd_soc_dapm_route intercon[] = {
437 437
438static int ak4671_add_widgets(struct snd_soc_codec *codec) 438static int ak4671_add_widgets(struct snd_soc_codec *codec)
439{ 439{
440 snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, 440 struct snd_soc_dapm_context *dapm = &codec->dapm;
441 ARRAY_SIZE(ak4671_dapm_widgets));
442 441
443 snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); 442 snd_soc_dapm_new_controls(dapm, ak4671_dapm_widgets,
443 ARRAY_SIZE(ak4671_dapm_widgets));
444 snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
444 445
445 return 0; 446 return 0;
446} 447}
@@ -602,7 +603,7 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec,
602 snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); 603 snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
603 break; 604 break;
604 } 605 }
605 codec->bias_level = level; 606 codec->dapm.bias_level = level;
606 return 0; 607 return 0;
607} 608}
608 609