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/tlv320aic3x.c | |
parent | 22e2fda5660cdf62513acabdb5c82a5af415f838 (diff) | |
download | linux-ce6120cca2589ede530200c7cfe11ac9f144333c.tar.bz2 |
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/tlv320aic3x.c')
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 22 |
1 files changed, 12 insertions, 10 deletions
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index fc687790188b..6173c2b4c364 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -183,7 +183,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { + list_for_each_entry(path, &widget->dapm->paths, list) { if (path->kcontrol != kcontrol) continue; @@ -199,7 +199,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync(widget->codec); + snd_soc_dapm_sync(widget->dapm); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); @@ -788,17 +788,19 @@ static const struct snd_soc_dapm_route intercon_3007[] = { static int aic3x_add_widgets(struct snd_soc_codec *codec) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_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)); if (aic3x->model == AIC3X_MODEL_3007) { - snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets, + snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); - snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007)); + snd_soc_dapm_add_routes(dapm, intercon_3007, + ARRAY_SIZE(intercon_3007)); } return 0; @@ -1135,7 +1137,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - if (codec->bias_level == SND_SOC_BIAS_STANDBY && + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); @@ -1146,7 +1148,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (!aic3x->power) aic3x_set_power(codec, 1); - if (codec->bias_level == SND_SOC_BIAS_PREPARE && + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); @@ -1159,7 +1161,7 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, aic3x_set_power(codec, 0); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1351,7 +1353,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) codec->control_data = aic3x->control_data; aic3x->codec = codec; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); if (ret != 0) { |