OpenCloudOS-Kernel/sound/soc/ti/omap-abe-twl6040.c

354 lines
9.3 KiB
C
Raw Normal View History

/*
* omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
* twl6040 codec
*
* Author: Misael Lopez Cruz <misael.lopez@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
* 02110-1301 USA
*
*/
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/mfd/twl6040.h>
#include <linux/module.h>
#include <linux/of.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "omap-dmic.h"
#include "omap-mcpdm.h"
#include "../codecs/twl6040.h"
struct abe_twl6040 {
struct snd_soc_card card;
struct snd_soc_dai_link dai_links[2];
int jack_detection; /* board can detect jack events */
int mclk_freq; /* MCLK frequency speed for twl6040 */
};
static struct platform_device *dmic_codec_dev;
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(codec_dai->component);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
freq = priv->mclk_freq;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
return -EINVAL;
/* set the codec mclk */
ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
SND_SOC_CLOCK_IN);
if (ret) {
printk(KERN_ERR "can't set codec system clock\n");
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
return ret;
}
static const struct snd_soc_ops omap_abe_ops = {
.hw_params = omap_abe_hw_params,
};
static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int ret = 0;
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC cpu system clock\n");
return ret;
}
ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
SND_SOC_CLOCK_OUT);
if (ret < 0) {
printk(KERN_ERR "can't set DMIC output clock\n");
return ret;
}
return 0;
}
static struct snd_soc_ops omap_abe_dmic_ops = {
.hw_params = omap_abe_dmic_hw_params,
};
/* Headset jack */
static struct snd_soc_jack hs_jack;
/*Headset jack detection DAPM pins */
static struct snd_soc_jack_pin hs_jack_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headset Stereophone",
.mask = SND_JACK_HEADPHONE,
},
};
/* SDP4430 machine DAPM */
static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
/* Outputs */
SND_SOC_DAPM_HP("Headset Stereophone", NULL),
SND_SOC_DAPM_SPK("Earphone Spk", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
SND_SOC_DAPM_SPK("Vibrator", NULL),
/* Inputs */
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
/* Digital microphones */
SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
/* Routings for outputs */
{"Headset Stereophone", NULL, "HSOL"},
{"Headset Stereophone", NULL, "HSOR"},
{"Earphone Spk", NULL, "EP"},
{"Ext Spk", NULL, "HFL"},
{"Ext Spk", NULL, "HFR"},
{"Line Out", NULL, "AUXL"},
{"Line Out", NULL, "AUXR"},
{"Vibrator", NULL, "VIBRAL"},
{"Vibrator", NULL, "VIBRAR"},
/* Routings for inputs */
{"HSMIC", NULL, "Headset Mic"},
{"Headset Mic", NULL, "Headset Mic Bias"},
{"MAINMIC", NULL, "Main Handset Mic"},
{"Main Handset Mic", NULL, "Main Mic Bias"},
{"SUBMIC", NULL, "Sub Handset Mic"},
{"Sub Handset Mic", NULL, "Main Mic Bias"},
{"AFML", NULL, "Line In"},
{"AFMR", NULL, "Line In"},
};
static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component = rtd->codec_dai->component;
struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
/*
* Configure McPDM offset cancellation based on the HSOTRIM value from
* twl6040.
*/
hs_trim = twl6040_get_trim_value(component, TWL6040_TRIM_HSOTRIM);
omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection only if it is supported */
if (priv->jack_detection) {
ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
SND_JACK_HEADSET, &hs_jack,
hs_jack_pins,
ARRAY_SIZE(hs_jack_pins));
if (ret)
return ret;
twl6040_hs_jack_detect(component, &hs_jack, SND_JACK_HEADSET);
}
return 0;
}
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic"},
{"Digital Mic", NULL, "Digital Mic1 Bias"},
};
static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
static int omap_abe_probe(struct platform_device *pdev)
{
struct device_node *node = pdev->dev.of_node;
struct snd_soc_card *card;
struct device_node *dai_node;
struct abe_twl6040 *priv;
int num_links = 0;
int ret = 0;
if (!node) {
dev_err(&pdev->dev, "of node is missing.\n");
return -ENODEV;
}
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
if (priv == NULL)
return -ENOMEM;
card = &priv->card;
card->dev = &pdev->dev;
card->owner = THIS_MODULE;
card->dapm_widgets = twl6040_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
card->dapm_routes = audio_map;
card->num_dapm_routes = ARRAY_SIZE(audio_map);
if (snd_soc_of_parse_card_name(card, "ti,model")) {
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
}
ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
if (ret) {
dev_err(&pdev->dev, "Error while parsing DAPM routing\n");
return ret;
}
dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
if (!dai_node) {
dev_err(&pdev->dev, "McPDM node is not provided\n");
return -EINVAL;
}
priv->dai_links[0].name = "DMIC";
priv->dai_links[0].stream_name = "TWL6040";
priv->dai_links[0].cpu_of_node = dai_node;
priv->dai_links[0].platform_of_node = dai_node;
priv->dai_links[0].codec_dai_name = "twl6040-legacy";
priv->dai_links[0].codec_name = "twl6040-codec";
priv->dai_links[0].init = omap_abe_twl6040_init;
priv->dai_links[0].ops = &omap_abe_ops;
dai_node = of_parse_phandle(node, "ti,dmic", 0);
if (dai_node) {
num_links = 2;
priv->dai_links[1].name = "TWL6040";
priv->dai_links[1].stream_name = "DMIC Capture";
priv->dai_links[1].cpu_of_node = dai_node;
priv->dai_links[1].platform_of_node = dai_node;
priv->dai_links[1].codec_dai_name = "dmic-hifi";
priv->dai_links[1].codec_name = "dmic-codec";
priv->dai_links[1].init = omap_abe_dmic_init;
priv->dai_links[1].ops = &omap_abe_dmic_ops;
} else {
num_links = 1;
}
priv->jack_detection = of_property_read_bool(node, "ti,jack-detection");
of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency not provided\n");
return -EINVAL;
}
card->fully_routed = 1;
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
return -ENODEV;
}
card->dai_link = priv->dai_links;
card->num_links = num_links;
snd_soc_card_set_drvdata(card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
static const struct of_device_id omap_abe_of_match[] = {
{.compatible = "ti,abe-twl6040", },
{ },
};
MODULE_DEVICE_TABLE(of, omap_abe_of_match);
static struct platform_driver omap_abe_driver = {
.driver = {
.name = "omap-abe-twl6040",
.pm = &snd_soc_pm_ops,
.of_match_table = omap_abe_of_match,
},
.probe = omap_abe_probe,
};
static int __init omap_abe_init(void)
{
int ret;
dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL,
0);
if (IS_ERR(dmic_codec_dev)) {
pr_err("%s: dmic-codec device registration failed\n", __func__);
return PTR_ERR(dmic_codec_dev);
}
ret = platform_driver_register(&omap_abe_driver);
if (ret) {
pr_err("%s: platform driver registration failed\n", __func__);
platform_device_unregister(dmic_codec_dev);
}
return ret;
}
module_init(omap_abe_init);
static void __exit omap_abe_exit(void)
{
platform_driver_unregister(&omap_abe_driver);
platform_device_unregister(dmic_codec_dev);
}
module_exit(omap_abe_exit);
MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:omap-abe-twl6040");