diff options
| author | Liam Girdwood <lrg@slimlogic.co.uk> | 2010-11-05 15:53:46 +0200 |
|---|---|---|
| committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2010-11-06 11:28:29 -0400 |
| commit | ce6120cca2589ede530200c7cfe11ac9f144333c (patch) | |
| tree | 6ea7c26ce64dd4753e7cf9a3b048e74614b169dc /sound/soc/codecs/tlv320aic23.c | |
| parent | 22e2fda5660cdf62513acabdb5c82a5af415f838 (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/tlv320aic23.c')
| -rw-r--r-- | sound/soc/codecs/tlv320aic23.c | 9 |
1 files changed, 5 insertions, 4 deletions
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index e8652b1ae326..d9d8e844d63f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -391,11 +391,12 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); /* set up audio path interconnects */ - snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); return 0; } @@ -574,7 +575,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } |