From ce6120cca2589ede530200c7cfe11ac9f144333c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 5 Nov 2010 15:53:46 +0200 Subject: 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 with some minor core changes, codecs and machine driver conversions from Jarkko Nikula . Signed-off-by: Liam Girdwood Signed-off-by: Jarkko Nikula Cc: Nicolas Ferre Cc: Manuel Lauss Cc: Mike Frysinger Cc: Cliff Cai Cc: Kevin Hilman Cc: Ryan Mallon Cc: Timur Tabi Cc: Sascha Hauer Cc: Lars-Peter Clausen Cc: Arnaud Patard (Rtp) Cc: Wan ZongShun Cc: Eric Miao Cc: Jassi Brar Cc: Daniel Gloeckner Cc: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) (limited to 'sound/soc/codecs/alc5623.c') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index fac61744f8c7..5a45067b43ba 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -832,7 +832,7 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } @@ -888,10 +888,10 @@ static int alc5623_resume(struct snd_soc_codec *codec) alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge alc5623 caps */ - if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->bias_level); + codec->dapm.bias_level = SND_SOC_BIAS_ON; + alc5623_set_bias_level(codec, codec->dapm.bias_level); } return 0; @@ -900,6 +900,7 @@ static int alc5623_resume(struct snd_soc_codec *codec) static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); @@ -943,24 +944,24 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, alc5623_snd_controls, ARRAY_SIZE(alc5623_snd_controls)); - snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets, + snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, ARRAY_SIZE(alc5623_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)); switch (alc5623->id) { default: case 0x21: case 0x22: - snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets, + snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, ARRAY_SIZE(alc5623_dapm_amp_widgets)); - snd_soc_dapm_add_routes(codec, intercon_amp_spk, - ARRAY_SIZE(intercon_amp_spk)); + snd_soc_dapm_add_routes(dapm, intercon_amp_spk, + ARRAY_SIZE(intercon_amp_spk)); break; case 0x23: - snd_soc_dapm_add_routes(codec, intercon_spk, - ARRAY_SIZE(intercon_spk)); + snd_soc_dapm_add_routes(dapm, intercon_spk, + ARRAY_SIZE(intercon_spk)); break; } -- cgit v1.2.1