[PATCH] ASoC: enable wm8753 in aspenite
Haojian Zhuang
haojian.zhuang at marvell.com
Fri Mar 19 18:13:32 EDT 2010
Enable wm8753 codec in aspenite via ssp interface.
Signed-off-by: Haojian Zhuang <haojian.zhuang at marvell.com>
---
sound/soc/pxa/Kconfig | 9 ++
sound/soc/pxa/Makefile | 2 +
sound/soc/pxa/aspenite.c | 218 ++++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 229 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/pxa/aspenite.c
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 286d52a..0f10ee2 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -113,6 +113,15 @@ config SND_PXA2XX_SOC_PALM27X
Say Y if you want to add support for SoC audio on
Palm T|X, T5, E2 or LifeDrive handheld computer.
+config SND_SOC_ASPENITE
+ tristate "SoC Audio support for Marvell Aspenite"
+ depends on SND_PXA2XX_SOC && MACH_ASPENITE
+ select SND_PXA168_SOC_SSP
+ select SND_SOC_WM8753
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Aspenite reference platform.
+
config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
depends on SND_PXA2XX_SOC && MACH_ZYLONITE
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index a74e6c9..74054aa 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -12,6 +12,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o
obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o
# PXA Machine Support
+snd-soc-aspenite-objs := aspenite.o
snd-soc-corgi-objs := corgi.o
snd-soc-poodle-objs := poodle.o
snd-soc-tosa-objs := tosa.o
@@ -38,6 +39,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
+obj-$(CONFIG_SND_SOC_ASPENITE) += snd-soc-aspenite.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
diff --git a/sound/soc/pxa/aspenite.c b/sound/soc/pxa/aspenite.c
new file mode 100644
index 0000000..733c95c
--- /dev/null
+++ b/sound/soc/pxa/aspenite.c
@@ -0,0 +1,218 @@
+/*
+ * aspenite.c -- SoC audio for Aspenite
+ *
+ * Copyright (C) 2009-2010 Marvell International Ltd.
+ * Haojian Zhuang <haojian.zhuang at marvell.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.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <plat/ssp.h>
+
+#include "../codecs/wm8753.h"
+#include "pxa2xx-pcm.h"
+#include "pxa168-ssp.h"
+
+static struct snd_soc_card aspenite;
+
+/* aspenite machine dapm widgets */
+static const struct snd_soc_dapm_widget aspenite_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_SPK("Headset Speaker L", NULL),
+ SND_SOC_DAPM_SPK("Headset Speaker R", NULL),
+ SND_SOC_DAPM_LINE("Line LIN", NULL),
+ SND_SOC_DAPM_LINE("Line RIN", NULL),
+};
+
+/* aspenite machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Headphone connected to LOUT1/ROUT1 */
+ {"Headphone", NULL, "LOUT1"},
+ {"Headphone", NULL, "ROUT1"},
+
+ /* Speaker connected to LOUT2/OUT4 & OUT3/ROUT2 */
+ {"Headset Speaker L", NULL, "LOUT2"},
+ {"Headset Speaker L", NULL, "OUT4"},
+ {"Headset Speaker R", NULL, "OUT3"},
+ {"Headset Speaker R", NULL, "ROUT2"},
+
+ /* Line connected to LINE1/LINE2 */
+ {"Line LIN", NULL, "LINE1"},
+ {"Line RIN", NULL, "LINE2"},
+
+ /* Mic */
+ {"MIC1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Mic"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+};
+
+static const struct snd_kcontrol_new wm8753_aspenite_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headset Speaker L"),
+ SOC_DAPM_PIN_SWITCH("Headset Speaker R"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Line LIN"),
+ SOC_DAPM_PIN_SWITCH("Line RIN"),
+};
+
+/* Seek the index of MCLK configuration table */
+static int pxa168_seek_mclk_conf(int rate, int format, int channel)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(mclk_conf); i++) {
+ if ((mclk_conf[i].rate == rate)
+ && (mclk_conf[i].format == format)
+ && (mclk_conf[i].channel == channel))
+ return i;
+ }
+ return -EINVAL;
+}
+
+/* Get the MCLK frequency */
+static int pxa168_get_mclk(int i)
+{
+ if ((i < 0) || (i >= ARRAY_SIZE(mclk_conf)))
+ return -EINVAL;
+ return mclk_conf[i].mclk;
+}
+
+static int aspenite_hifi_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->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int rate, width, channel;
+ int index, mclk, ret;
+
+ rate = params_rate(params);
+ width = snd_pcm_format_physical_width(params_format(params));
+ channel = params_channels(params);
+ ret = pxa168_seek_mclk_conf(rate, width, channel);
+ if (ret < 0)
+ return ret;
+ index = ret;
+ mclk = pxa168_get_mclk(ret);
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu system clock */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, index,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops aspenite_hifi_ops = {
+ .hw_params = aspenite_hifi_hw_params,
+};
+
+static int aspenite_wm8753_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ /* set up NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "MONO1");
+ snd_soc_dapm_nc_pin(codec, "MONO2");
+ snd_soc_dapm_nc_pin(codec, "RXP");
+ snd_soc_dapm_nc_pin(codec, "RXN");
+
+ snd_soc_dapm_new_controls(codec, aspenite_dapm_widgets,
+ ARRAY_SIZE(aspenite_dapm_widgets));
+ ret = snd_soc_add_controls(codec, wm8753_aspenite_controls,
+ ARRAY_SIZE(wm8753_aspenite_controls));
+ if (ret < 0)
+ return ret;
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ return 0;
+}
+
+static struct snd_soc_dai_link aspenite_dai[] = {
+ {
+ .name = "WM8753 HiFi",
+ .stream_name = "WM8753 HiFi",
+ .cpu_dai = &pxa168_ssp_dai[PXA168_DAI_SSP1],
+ .codec_dai = &wm8753_dai[0],
+ .init = aspenite_wm8753_init,
+ .ops = &aspenite_hifi_ops,
+ },
+};
+
+static struct snd_soc_card aspenite = {
+ .name = "Aspenite",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = aspenite_dai,
+ .num_links = ARRAY_SIZE(aspenite_dai),
+};
+
+static struct snd_soc_device aspenite_snd_devdata = {
+ .card = &aspenite,
+ .codec_dev = &soc_codec_dev_wm8753,
+};
+
+static struct platform_device *aspenite_snd_device;
+
+static int __init aspenite_init(void)
+{
+ int ret;
+
+ aspenite_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!aspenite_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(aspenite_snd_device, &aspenite_snd_devdata);
+ aspenite_snd_devdata.dev = &aspenite_snd_device->dev;
+
+ ret = platform_device_add(aspenite_snd_device);
+ if (ret)
+ platform_device_put(aspenite_snd_device);
+ return ret;
+}
+module_init(aspenite_init);
+
+static void __exit aspenite_exit(void)
+{
+ platform_device_unregister(aspenite_snd_device);
+}
+module_exit(aspenite_exit);
+
+MODULE_DESCRIPTION("ALSA SoC WM8753 Aspenite");
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang at marvell.com>");
+MODULE_LICENSE("GPL");
--
1.5.6.5
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