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authorLiam Girdwood <lrg@slimlogic.co.uk>2010-11-05 15:53:46 +0200
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-11-06 11:28:29 -0400
commitce6120cca2589ede530200c7cfe11ac9f144333c (patch)
tree6ea7c26ce64dd4753e7cf9a3b048e74614b169dc /sound/soc/atmel
parent22e2fda5660cdf62513acabdb5c82a5af415f838 (diff)
downloadop-kernel-dev-ce6120cca2589ede530200c7cfe11ac9f144333c.zip
op-kernel-dev-ce6120cca2589ede530200c7cfe11ac9f144333c.tar.gz
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/atmel')
-rw-r--r--sound/soc/atmel/playpaq_wm8510.c11
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c17
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c13
3 files changed, 22 insertions, 19 deletions
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 5f4e59f..aede7e7 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -318,27 +318,28 @@ static const struct snd_soc_dapm_route intercon[] = {
static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int i;
/*
* Add DAPM widgets
*/
for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
- snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]);
+ snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
/*
* Setup audio path interconnects
*/
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
/* always connected pins */
- snd_soc_dapm_enable_pin(codec, "Int Mic");
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_enable_pin(dapm, "Int Mic");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_sync(dapm);
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569d..da9c303 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -140,6 +140,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
printk(KERN_DEBUG
@@ -154,25 +155,25 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
}
/* Add specific widgets */
- snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets,
ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
/* Set up specific audio path interconnects */
- snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
/* not connected */
- snd_soc_dapm_nc_pin(codec, "RLINEIN");
- snd_soc_dapm_nc_pin(codec, "LLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "RLINEIN");
+ snd_soc_dapm_nc_pin(dapm, "LLINEIN");
#ifdef ENABLE_MIC_INPUT
- snd_soc_dapm_enable_pin(codec, "Int Mic");
+ snd_soc_dapm_enable_pin(dapm, "Int Mic");
#else
- snd_soc_dapm_nc_pin(codec, "Int Mic");
+ snd_soc_dapm_nc_pin(dapm, "Int Mic");
#endif
/* always connected */
- snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d2835..92c709e 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -105,19 +105,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Add afeb9260 specific widgets */
- snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* Set up afeb9260 specific audio path audio_map */
- snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- snd_soc_dapm_enable_pin(codec, "Headphone Jack");
- snd_soc_dapm_enable_pin(codec, "Line In");
- snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Line In");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_sync(codec);
+ snd_soc_dapm_sync(dapm);
return 0;
}
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