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-rw-r--r--sound/soc/omap/Kconfig8
-rw-r--r--sound/soc/omap/Makefile2
-rw-r--r--sound/soc/omap/n810.c6
-rw-r--r--sound/soc/omap/omap-mcbsp.c181
-rw-r--r--sound/soc/omap/omap-mcbsp.h16
-rw-r--r--sound/soc/omap/omap-pcm.c4
-rw-r--r--sound/soc/omap/osk5912.c232
7 files changed, 386 insertions, 63 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index aea27e70043..8b7766b998d 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_OSK5912
+ tristate "SoC Audio support for omap osk5912"
+ depends on SND_OMAP_SOC && MACH_OMAP_OSK
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TLV320AIC23
+ help
+ Say Y if you want to add support for SoC audio on osk5912.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index d8d8d58075e..e09d1f297f6 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
+snd-soc-osk5912-objs := osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index d166b6b2a60..fae3ad36e0b 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
int i, err;
/* Not connected */
- snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
- snd_soc_dapm_disable_pin(codec, "HPLCOM");
- snd_soc_dapm_disable_pin(codec, "HPRCOM");
+ snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+ snd_soc_dapm_nc_pin(codec, "HPLCOM");
+ snd_soc_dapm_nc_pin(codec, "HPRCOM");
/* Add N810 specific controls */
for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 35310e16d7f..0a063a98a66 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -59,12 +59,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
* Stream DMA parameters. DMA request line and port address are set runtime
* since they are different between OMAP1 and later OMAPs
*/
-static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
-{
- { .name = "I2S PCM Stereo out", },
- { .name = "I2S PCM Stereo in", },
-},
-};
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
static const int omap1_dma_reqs[][2] = {
@@ -84,11 +79,22 @@ static const unsigned long omap1_mcbsp_port[][2] = {
static const int omap1_dma_reqs[][2] = {};
static const unsigned long omap1_mcbsp_port[][2] = {};
#endif
-#if defined(CONFIG_ARCH_OMAP2420)
-static const int omap2420_dma_reqs[][2] = {
+
+#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
+static const int omap24xx_dma_reqs[][2] = {
{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+ { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
+ { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
+ { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
+#endif
};
+#else
+static const int omap24xx_dma_reqs[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP2420)
static const unsigned long omap2420_mcbsp_port[][2] = {
{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
@@ -96,10 +102,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = {
OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
};
#else
-static const int omap2420_dma_reqs[][2] = {};
static const unsigned long omap2420_mcbsp_port[][2] = {};
#endif
+#if defined(CONFIG_ARCH_OMAP2430)
+static const unsigned long omap2430_mcbsp_port[][2] = {
+ { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap2430_mcbsp_port[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP34XX)
+static const unsigned long omap34xx_mcbsp_port[][2] = {
+ { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap34xx_mcbsp_port[][2] = {};
+#endif
+
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -167,14 +206,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
dma = omap1_dma_reqs[bus_id][substream->stream];
port = omap1_mcbsp_port[bus_id][substream->stream];
} else if (cpu_is_omap2420()) {
- dma = omap2420_dma_reqs[bus_id][substream->stream];
+ dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap2420_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap2430()) {
+ dma = omap24xx_dma_reqs[bus_id][substream->stream];
+ port = omap2430_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap343x()) {
+ dma = omap24xx_dma_reqs[bus_id][substream->stream];
+ port = omap34xx_mcbsp_port[bus_id][substream->stream];
} else {
- /*
- * TODO: Add support for 2430 and 3430
- */
return -ENODEV;
}
+ omap_mcbsp_dai_dma_params[id][substream->stream].name =
+ substream->stream ? "Audio Capture" : "Audio Playback";
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
@@ -245,6 +289,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* 0-bit data delay */
+ regs->rcr2 |= RDATDLY(0);
+ regs->xcr2 |= XDATDLY(0);
+ break;
default:
/* Unsupported data format */
return -EINVAL;
@@ -310,7 +359,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
int clk_id)
{
int sel_bit;
- u16 reg;
+ u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1;
if (cpu_class_is_omap1()) {
/* OMAP1's can use only external source clock */
@@ -320,6 +369,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
return 0;
}
+ if (cpu_is_omap2420() && mcbsp_data->bus_id > 1)
+ return -EINVAL;
+
+ if (cpu_is_omap343x())
+ reg_devconf1 = OMAP343X_CONTROL_DEVCONF1;
+
switch (mcbsp_data->bus_id) {
case 0:
reg = OMAP2_CONTROL_DEVCONF0;
@@ -329,20 +384,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
reg = OMAP2_CONTROL_DEVCONF0;
sel_bit = 6;
break;
- /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
+ case 2:
+ reg = reg_devconf1;
+ sel_bit = 0;
+ break;
+ case 3:
+ reg = reg_devconf1;
+ sel_bit = 2;
+ break;
+ case 4:
+ reg = reg_devconf1;
+ sel_bit = 4;
+ break;
default:
return -EINVAL;
}
- if (cpu_class_is_omap2()) {
- if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
- omap_ctrl_writel(omap_ctrl_readl(reg) &
- ~(1 << sel_bit), reg);
- } else {
- omap_ctrl_writel(omap_ctrl_readl(reg) |
- (1 << sel_bit), reg);
- }
- }
+ if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)
+ omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
+ else
+ omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
return 0;
}
@@ -376,37 +437,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
return err;
}
-struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
-{
- .name = "omap-mcbsp-dai",
- .id = 0,
- .type = SND_SOC_DAI_I2S,
- .playback = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = OMAP_MCBSP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
- .capture = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = OMAP_MCBSP_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- },
- .ops = {
- .startup = omap_mcbsp_dai_startup,
- .shutdown = omap_mcbsp_dai_shutdown,
- .trigger = omap_mcbsp_dai_trigger,
- .hw_params = omap_mcbsp_dai_hw_params,
- },
- .dai_ops = {
- .set_fmt = omap_mcbsp_dai_set_dai_fmt,
- .set_clkdiv = omap_mcbsp_dai_set_clkdiv,
- .set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
- },
- .private_data = &mcbsp_data[0].bus_id,
-},
+#define OMAP_MCBSP_DAI_BUILDER(link_id) \
+{ \
+ .name = "omap-mcbsp-dai-(link_id)", \
+ .id = (link_id), \
+ .type = SND_SOC_DAI_I2S, \
+ .playback = { \
+ .channels_min = 2, \
+ .channels_max = 2, \
+ .rates = OMAP_MCBSP_RATES, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ }, \
+ .capture = { \
+ .channels_min = 2, \
+ .channels_max = 2, \
+ .rates = OMAP_MCBSP_RATES, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ }, \
+ .ops = { \
+ .startup = omap_mcbsp_dai_startup, \
+ .shutdown = omap_mcbsp_dai_shutdown, \
+ .trigger = omap_mcbsp_dai_trigger, \
+ .hw_params = omap_mcbsp_dai_hw_params, \
+ }, \
+ .dai_ops = { \
+ .set_fmt = omap_mcbsp_dai_set_dai_fmt, \
+ .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \
+ .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \
+ }, \
+ .private_data = &mcbsp_data[(link_id)].bus_id, \
+}
+
+struct snd_soc_dai omap_mcbsp_dai[] = {
+ OMAP_MCBSP_DAI_BUILDER(0),
+ OMAP_MCBSP_DAI_BUILDER(1),
+#if NUM_LINKS >= 3
+ OMAP_MCBSP_DAI_BUILDER(2),
+#endif
+#if NUM_LINKS == 5
+ OMAP_MCBSP_DAI_BUILDER(3),
+ OMAP_MCBSP_DAI_BUILDER(4),
+#endif
};
+
EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index ed8afb55067..df7ad13ba73 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -38,11 +38,17 @@ enum omap_mcbsp_div {
OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */
};
-/*
- * REVISIT: Preparation for the ASoC v2. Let the number of available links to
- * be same than number of McBSP ports found in OMAP(s) we are compiling for.
- */
-#define NUM_LINKS 1
+#if defined(CONFIG_ARCH_OMAP2420)
+#define NUM_LINKS 2
+#endif
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+#undef NUM_LINKS
+#define NUM_LINKS 3
+#endif
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#undef NUM_LINKS
+#define NUM_LINKS 5
+#endif
extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 690bfeaec4a..e9084fdd208 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
prtd->dma_data = dma_data;
err = omap_request_dma(dma_data->dma_req, dma_data->name,
omap_pcm_dma_irq, substream, &prtd->dma_ch);
- if (!cpu_is_omap1510()) {
+ if (!err & !cpu_is_omap1510()) {
/*
* Link channel with itself so DMA doesn't need any
* reprogramming while looping the buffer
@@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC;
dma_params.src_start = runtime->dma_addr;
dma_params.dst_start = dma_data->port_addr;
+ dma_params.dst_port = OMAP_DMA_PORT_MPUI;
} else {
dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT;
dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
dma_params.src_start = dma_data->port_addr;
dma_params.dst_start = runtime->dma_addr;
+ dma_params.src_port = OMAP_DMA_PORT_MPUI;
}
/*
* Set DMA transfer frame size equal to ALSA period size and frame
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 00000000000..0fe73379689
--- /dev/null
+++ b/sound/soc/omap/osk5912.c
@@ -0,0 +1,232 @@
+/*
+ * osk5912.c -- SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS <arunks@mistralsolutions.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 <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+ return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+ clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_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;
+ int err;
+
+ /* Set codec DAI configuration */
+ err = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return err;
+ }
+
+ /* Set cpu DAI configuration */
+ err = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return err;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ err =
+ snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return err;
+ }
+
+ return err;
+}
+
+static struct snd_soc_ops osk_ops = {
+ .startup = osk_startup,
+ .hw_params = osk_hw_params,
+ .shutdown = osk_shutdown,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ {"LLINEIN", NULL, "Line In"},
+ {"RLINEIN", NULL, "Line In"},
+
+ {"MICIN", NULL, "Mic Jack"},
+};
+
+static int osk_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+ /* Add osk5912 specific widgets */
+ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+ /* Set up osk5912 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, 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_sync(codec);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link osk_dai = {
+ .name = "TLV320AIC23",
+ .stream_name = "AIC23",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &tlv320aic23_dai,
+ .init = osk_tlv320aic23_init,
+ .ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_osk = {
+ .name = "OSK5912",
+ .dai_link = &osk_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+ .machine = &snd_soc_machine_osk,
+ .platform = &omap_soc_platform,
+ .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+ int err;
+ u32 curRate;
+ struct device *dev;
+
+ if (!(machine_is_omap_osk()))
+ return -ENODEV;
+
+ osk_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!osk_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
+ osk_snd_devdata.dev = &osk_snd_device->dev;
+ *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */
+ err = platform_device_add(osk_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &osk_snd_device->dev;
+
+ tlv320aic23_mclk = clk_get(dev, "mclk");
+ if (IS_ERR(tlv320aic23_mclk)) {
+ printk(KERN_ERR "Could not get mclk clock\n");
+ return -ENODEV;
+ }
+
+ if (clk_get_usecount(tlv320aic23_mclk) > 0) {
+ /* MCLK is already in use */
+ printk(KERN_WARNING
+ "MCLK in use at %d Hz. We change it to %d Hz\n",
+ (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+ }
+
+ /*
+ * Configure 12 MHz output on MCLK.
+ */
+ curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+ if (curRate != CODEC_CLOCK) {
+ if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+ printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+ err = -ECANCELED;
+ goto err1;
+ }
+ }
+
+ printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
+ (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
+ clk_get_usecount(tlv320aic23_mclk));
+
+ return 0;
+err1:
+ clk_put(tlv320aic23_mclk);
+ platform_device_del(osk_snd_device);
+ platform_device_put(osk_snd_device);
+
+ return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+ platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-24 22:18:06 +0000