diff options
Diffstat (limited to 'sound/soc/omap')
-rw-r--r-- | sound/soc/omap/Kconfig | 13 | ||||
-rw-r--r-- | sound/soc/omap/Makefile | 4 | ||||
-rw-r--r-- | sound/soc/omap/omap-abe-twl6040.c | 349 | ||||
-rw-r--r-- | sound/soc/omap/sdp4430.c | 279 |
4 files changed, 359 insertions, 286 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index fb1bf2581ef..47b23fea20c 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -97,16 +97,19 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. -config SND_OMAP_SOC_SDP4430 - tristate "SoC Audio support for Texas Instruments SDP4430" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP +config SND_OMAP_SOC_OMAP_ABE_TWL6040 + tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" + depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4 select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC help - Say Y if you want to add support for SoC audio on Texas Instruments - SDP4430. + Say Y if you want to add support for SoC audio on OMAP boards using + ABE and twl6040 codec. This driver currently supports: + - SDP4430/Blaze boards + - PandaBoard (4430) + - PandaBoardES (4460) config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 1fd723fb559..123ac18303e 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -20,7 +20,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o -snd-soc-sdp4430-objs := sdp4430.o +snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -36,7 +36,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o -obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c new file mode 100644 index 00000000000..93bb8eee22b --- /dev/null +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -0,0 +1,349 @@ +/* + * 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/platform_data/omap-abe-twl6040.h> +#include <linux/module.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> +#include <plat/hardware.h> +#include <plat/mux.h> + +#include "omap-dmic.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" +#include "../codecs/twl6040.h" + +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; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); + int clk_id, freq; + int ret; + + clk_id = twl6040_get_clk_id(rtd->codec); + if (clk_id == TWL6040_SYSCLK_SEL_HPPLL) + freq = pdata->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; + } + return ret; +} + +static 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), +}; + +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 inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, + int connected, char *pin) +{ + if (!connected) + snd_soc_dapm_disable_pin(dapm, pin); +} + +static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); + int hs_trim; + int ret = 0; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); + twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); + twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); + + /* + * Configure McPDM offset cancellation based on the HSOTRIM value from + * twl6040. + */ + hs_trim = twl6040_get_trim_value(codec, 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 (pdata->jack_detection) { + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); + } + + return ret; +} + +static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Digital Mic", NULL), +}; + +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_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, + ARRAY_SIZE(dmic_dapm_widgets)); + if (ret) + return ret; + + return snd_soc_dapm_add_routes(dapm, dmic_audio_map, + ARRAY_SIZE(dmic_audio_map)); +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link twl6040_dmic_dai[] = { + { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai_name = "omap-mcpdm", + .codec_dai_name = "twl6040-legacy", + .platform_name = "omap-pcm-audio", + .codec_name = "twl6040-codec", + .init = omap_abe_twl6040_init, + .ops = &omap_abe_ops, + }, + { + .name = "DMIC", + .stream_name = "DMIC Capture", + .cpu_dai_name = "omap-dmic", + .codec_dai_name = "dmic-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "dmic-codec", + .init = omap_abe_dmic_init, + .ops = &omap_abe_dmic_ops, + }, +}; + +static struct snd_soc_dai_link twl6040_only_dai[] = { + { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai_name = "omap-mcpdm", + .codec_dai_name = "twl6040-legacy", + .platform_name = "omap-pcm-audio", + .codec_name = "twl6040-codec", + .init = omap_abe_twl6040_init, + .ops = &omap_abe_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card omap_abe_card = { + .owner = THIS_MODULE, + + .dapm_widgets = twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), +}; + +static __devinit int omap_abe_probe(struct platform_device *pdev) +{ + struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct snd_soc_card *card = &omap_abe_card; + int ret; + + card->dev = &pdev->dev; + + if (!pdata) { + dev_err(&pdev->dev, "Missing pdata\n"); + return -ENODEV; + } + + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + if (!pdata->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency missing\n"); + return -ENODEV; + } + + if (pdata->has_dmic) { + card->dai_link = twl6040_dmic_dai; + card->num_links = ARRAY_SIZE(twl6040_dmic_dai); + } else { + card->dai_link = twl6040_only_dai; + card->num_links = ARRAY_SIZE(twl6040_only_dai); + } + + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + + return ret; +} + +static int __devexit omap_abe_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver omap_abe_driver = { + .driver = { + .name = "omap-abe-twl6040", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = omap_abe_probe, + .remove = __devexit_p(omap_abe_remove), +}; + +module_platform_driver(omap_abe_driver); + +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"); diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c deleted file mode 100644 index 175ba9a04ed..00000000000 --- a/sound/soc/omap/sdp4430.c +++ /dev/null @@ -1,279 +0,0 @@ -/* - * sdp4430.c -- SoC audio for TI OMAP4430 SDP - * - * Author: Misael Lopez Cruz <x0052729@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 <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/jack.h> - -#include <asm/mach-types.h> -#include <plat/hardware.h> -#include <plat/mux.h> - -#include "omap-dmic.h" -#include "omap-mcpdm.h" -#include "omap-pcm.h" -#include "../codecs/twl6040.h" - -static int sdp4430_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 *codec_dai = rtd->codec_dai; - int clk_id, freq; - int ret; - - clk_id = twl6040_get_clk_id(rtd->codec); - if (clk_id == TWL6040_SYSCLK_SEL_HPPLL) - freq = 38400000; - 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; - } - return ret; -} - -static struct snd_soc_ops sdp4430_ops = { - .hw_params = sdp4430_hw_params, -}; - -static int sdp4430_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 sdp4430_dmic_ops = { - .hw_params = sdp4430_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 sdp4430_twl6040_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Ext Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("FM Stereo In"), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias*/ - {"MAINMIC", NULL, "Main Mic Bias"}, - {"SUBMIC", NULL, "Main Mic Bias"}, - {"Main Mic Bias", NULL, "Ext Mic"}, - - /* External Speakers: HFL, HFR */ - {"Ext Spk", NULL, "HFL"}, - {"Ext Spk", NULL, "HFR"}, - - /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Mic"}, - - /* Headset Stereophone (Headphone): HSOL, HSOR */ - {"Headset Stereophone", NULL, "HSOL"}, - {"Headset Stereophone", NULL, "HSOR"}, - - /* Earphone speaker */ - {"Earphone Spk", NULL, "EP"}, - - /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "FM Stereo In"}, - {"AFMR", NULL, "FM Stereo In"}, -}; - -static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - int ret, hs_trim; - - /* - * Configure McPDM offset cancellation based on the HSOTRIM value from - * twl6040. - */ - hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM); - omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim), - TWL6040_HSF_TRIM_RIGHT(hs_trim)); - - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - - if (machine_is_omap_4430sdp()) - twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); - else - snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); - - return ret; -} - -static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Digital Mic", NULL), -}; - -static const struct snd_soc_dapm_route dmic_audio_map[] = { - {"DMic", NULL, "Digital Mic1 Bias"}, - {"Digital Mic1 Bias", NULL, "Digital Mic"}, -}; - -static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, - ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); - if (ret) - return ret; - - return snd_soc_dapm_add_routes(dapm, dmic_audio_map, - ARRAY_SIZE(dmic_audio_map)); -} - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sdp4430_dai[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .cpu_dai_name = "omap-mcpdm", - .codec_dai_name = "twl6040-legacy", - .platform_name = "omap-pcm-audio", - .codec_name = "twl6040-codec", - .init = sdp4430_twl6040_init, - .ops = &sdp4430_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .cpu_dai_name = "omap-dmic", - .codec_dai_name = "dmic-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "dmic-codec", - .init = sdp4430_dmic_init, - .ops = &sdp4430_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_sdp4430 = { - .name = "SDP4430", - .owner = THIS_MODULE, - .dai_link = sdp4430_dai, - .num_links = ARRAY_SIZE(sdp4430_dai), - - .dapm_widgets = sdp4430_twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *sdp4430_snd_device; - -static int __init sdp4430_soc_init(void) -{ - int ret; - - if (!machine_is_omap_4430sdp()) - return -ENODEV; - printk(KERN_INFO "SDP4430 SoC init\n"); - - sdp4430_snd_device = platform_device_alloc("soc-audio", -1); - if (!sdp4430_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); - - ret = platform_device_add(sdp4430_snd_device); - if (ret) - goto err; - - return 0; - -err: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(sdp4430_snd_device); - return ret; -} -module_init(sdp4430_soc_init); - -static void __exit sdp4430_soc_exit(void) -{ - platform_device_unregister(sdp4430_snd_device); -} -module_exit(sdp4430_soc_exit); - -MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>"); -MODULE_DESCRIPTION("ALSA SoC SDP4430"); -MODULE_LICENSE("GPL"); - |