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/tlv320dac33.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/tlv320dac33.c')
| -rw-r--r-- | sound/soc/codecs/tlv320dac33.c | 15 |
1 files changed, 8 insertions, 7 deletions
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c5ab8c805771..7149c14b289e 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -628,11 +628,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int dac33_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, - ARRAY_SIZE(dac33_dapm_widgets)); + struct snd_soc_dapm_context *dapm = &codec->dapm; + snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); /* set up 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)); return 0; } @@ -649,7 +650,7 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->bias_level == SND_SOC_BIAS_OFF) { + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Coming from OFF, switch on the codec */ ret = dac33_hard_power(codec, 1); if (ret != 0) @@ -660,14 +661,14 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: /* Do not power off, when the codec is already off */ - if (codec->bias_level == SND_SOC_BIAS_OFF) + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) return 0; ret = dac33_hard_power(codec, 0); if (ret != 0) return ret; break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -1415,7 +1416,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->idle_bias_off = 1; + codec->dapm.idle_bias_off = 1; dac33->codec = codec; /* Read the tlv320dac33 ID registers */ |