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-rw-r--r--sound/soc/omap/Kconfig43
-rw-r--r--sound/soc/omap/Makefile10
-rw-r--r--sound/soc/omap/n810.c16
-rw-r--r--sound/soc/omap/omap-mcbsp.c239
-rw-r--r--sound/soc/omap/omap-mcbsp.h16
-rw-r--r--sound/soc/omap/omap-pcm.c18
-rw-r--r--sound/soc/omap/omap2evm.c151
-rw-r--r--sound/soc/omap/omap3beagle.c149
-rw-r--r--sound/soc/omap/omap3pandora.c311
-rw-r--r--sound/soc/omap/osk5912.c232
-rw-r--r--sound/soc/omap/overo.c148
-rw-r--r--sound/soc/omap/sdp3430.c152
12 files changed, 1403 insertions, 82 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index aea27e70043..a7b1d77b210 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,6 +1,6 @@
config SND_OMAP_SOC
tristate "SoC Audio for the Texas Instruments OMAP chips"
- depends on ARCH_OMAP && SND_SOC
+ depends on ARCH_OMAP
config SND_OMAP_SOC_MCBSP
tristate
@@ -13,3 +13,44 @@ 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.
+
+config SND_OMAP_SOC_OVERO
+ tristate "SoC Audio support for Gumstix Overo"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the Gumstix Overo.
+
+config SND_OMAP_SOC_OMAP2EVM
+ tristate "SoC Audio support for OMAP2EVM board"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the omap2evm board.
+
+config SND_OMAP_SOC_SDP3430
+ tristate "SoC Audio support for Texas Instruments SDP3430"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on Texas Instruments
+ SDP3430.
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+ tristate "SoC Audio support for OMAP3 Pandora"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index d8d8d58075e..76fedd96e36 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,5 +7,15 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
+snd-soc-osk5912-objs := osk5912.o
+snd-soc-overo-objs := overo.o
+snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-omap3pandora-objs := omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
+obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index d166b6b2a60..25593fee912 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -70,9 +70,13 @@ static void n810_ext_control(struct snd_soc_codec *codec)
static int n810_startup(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->socdev->codec;
+ snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+
n810_ext_control(codec);
return clk_enable(sys_clkout2);
}
@@ -247,9 +251,9 @@ 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++) {
@@ -282,8 +286,9 @@ static struct snd_soc_dai_link n810_dai = {
};
/* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_n810 = {
+static struct snd_soc_card snd_soc_n810 = {
.name = "N810",
+ .platform = &omap_soc_platform,
.dai_link = &n810_dai,
.num_links = 1,
};
@@ -298,8 +303,7 @@ static struct aic3x_setup_data n810_aic33_setup = {
/* Audio subsystem */
static struct snd_soc_device n810_snd_devdata = {
- .machine = &snd_soc_machine_n810,
- .platform = &omap_soc_platform,
+ .card = &snd_soc_n810,
.codec_dev = &soc_codec_dev_aic3x,
.codec_data = &n810_aic33_setup,
};
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 35310e16d7f..ec5e18a7875 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -36,13 +36,12 @@
#include "omap-mcbsp.h"
#include "omap-pcm.h"
-#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \
- SNDRV_PCM_RATE_48000 | \
- SNDRV_PCM_RATE_KNOT)
+#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000)
struct omap_mcbsp_data {
unsigned int bus_id;
struct omap_mcbsp_reg_cfg regs;
+ unsigned int fmt;
/*
* Flags indicating is the bus already activated and configured by
* another substream
@@ -59,12 +58,7 @@ 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 +78,22 @@ 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 },
@@ -96,11 +101,45 @@ 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
-static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+#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_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -113,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
return err;
}
-static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -125,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
}
}
-static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -154,27 +195,34 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
}
static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+ int wlen, channels;
unsigned long port;
if (cpu_class_is_omap1()) {
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];
@@ -184,12 +232,17 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
return 0;
}
- switch (params_channels(params)) {
+ channels = params_channels(params);
+ switch (channels) {
case 2:
- /* Set 1 word per (McBPSP) frame and use dual-phase frames */
- regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE;
+ /* Use dual-phase frames */
+ regs->rcr2 |= RPHASE;
+ regs->xcr2 |= XPHASE;
+ case 1:
+ /* Set 1 word per (McBSP) frame */
+ regs->rcr2 |= RFRLEN2(1 - 1);
regs->rcr1 |= RFRLEN1(1 - 1);
- regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE;
+ regs->xcr2 |= XFRLEN2(1 - 1);
regs->xcr1 |= XFRLEN1(1 - 1);
break;
default:
@@ -200,19 +253,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
/* Set word lengths */
+ wlen = 16;
regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16);
regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16);
regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16);
regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16);
- /* Set FS period and length in terms of bit clock periods */
- regs->srgr2 |= FPER(16 * 2 - 1);
- regs->srgr1 |= FWID(16 - 1);
break;
default:
/* Unsupported PCM format */
return -EINVAL;
}
+ /* Set FS period and length in terms of bit clock periods */
+ switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr1 |= FWID(wlen - 1);
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ regs->srgr2 |= FPER(wlen * channels - 1);
+ regs->srgr1 |= FWID(wlen * channels - 2);
+ break;
+ }
+
omap_mcbsp_config(bus_id, &mcbsp_data->regs);
mcbsp_data->configured = 1;
@@ -232,6 +295,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
if (mcbsp_data->configured)
return 0;
+ mcbsp_data->fmt = fmt;
memset(regs, 0, sizeof(*regs));
/* Generic McBSP register settings */
regs->spcr2 |= XINTM(3) | FREE;
@@ -245,6 +309,11 @@ 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_B:
+ /* 0-bit data delay */
+ regs->rcr2 |= RDATDLY(0);
+ regs->xcr2 |= XDATDLY(0);
+ break;
default:
/* Unsupported data format */
return -EINVAL;
@@ -310,7 +379,7 @@ 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 +389,12 @@ 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 +404,26 @@ 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;
}
@@ -376,39 +457,61 @@ 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), \
+ .playback = { \
+ .channels_min = 1, \
+ .channels_max = 2, \
+ .rates = OMAP_MCBSP_RATES, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ }, \
+ .capture = { \
+ .channels_min = 1, \
+ .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, \
+ .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);
+static int __init snd_omap_mcbsp_init(void)
+{
+ return snd_soc_register_dais(omap_mcbsp_dai,
+ ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_init(snd_omap_mcbsp_init);
+
+static void __exit snd_omap_mcbsp_exit(void)
+{
+ snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_exit(snd_omap_mcbsp_exit);
+
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
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..b0362dfd5b7 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
@@ -231,7 +233,7 @@ static int omap_pcm_open(struct snd_pcm_substream *substream)
if (ret < 0)
goto out;
- prtd = kzalloc(sizeof(prtd), GFP_KERNEL);
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
if (prtd == NULL) {
ret = -ENOMEM;
goto out;
@@ -352,6 +354,18 @@ struct snd_soc_platform omap_soc_platform = {
};
EXPORT_SYMBOL_GPL(omap_soc_platform);
+static int __init omap_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&omap_soc_platform);
+}
+module_init(omap_soc_platform_init);
+
+static void __exit omap_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&omap_soc_platform);
+}
+module_exit(omap_soc_platform_exit);
+
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
new file mode 100644
index 00000000000..0c2322dcf02
--- /dev/null
+++ b/sound/soc/omap/omap2evm.c
@@ -0,0 +1,151 @@
+/*
+ * omap2evm.c -- SoC audio machine driver for omap2evm board
+ *
+ * Author: Arun KS <arunks@mistralsolutions.com>
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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 <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap2evm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ 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 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap2evm_ops = {
+ .hw_params = omap2evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap2evm_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap2evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap2evm = {
+ .name = "omap2evm",
+ .platform = &omap_soc_platform,
+ .dai_link = &omap2evm_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap2evm_snd_devdata = {
+ .card = &snd_soc_omap2evm,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap2evm_snd_device;
+
+static int __init omap2evm_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap2evm()) {
+ pr_debug("Not omap2evm!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "omap2evm SoC init\n");
+
+ omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!omap2evm_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata);
+ omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev;
+ *(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(omap2evm_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(omap2evm_snd_device);
+
+ return ret;
+}
+module_init(omap2evm_soc_init);
+
+static void __exit omap2evm_soc_exit(void)
+{
+ platform_device_unregister(omap2evm_snd_device);
+}
+module_exit(omap2evm_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC omap2evm");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
new file mode 100644
index 00000000000..fd24a4acd2f
--- /dev/null
+++ b/sound/soc/omap/omap3beagle.c
@@ -0,0 +1,149 @@
+/*
+ * omap3beagle.c -- SoC audio for OMAP3 Beagle
+ *
+ * Author: Steve Sakoman <steve@sakoman.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 <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3beagle_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 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap3beagle_ops = {
+ .hw_params = omap3beagle_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3beagle_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap3beagle_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3beagle = {
+ .name = "omap3beagle",
+ .platform = &omap_soc_platform,
+ .dai_link = &omap3beagle_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3beagle_snd_devdata = {
+ .card = &snd_soc_omap3beagle,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3beagle_snd_device;
+
+static int __init omap3beagle_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap3_beagle()) {
+ pr_debug("Not OMAP3 Beagle!\n");
+ return -ENODEV;
+ }
+ pr_info("OMAP3 Beagle SoC init\n");
+
+ omap3beagle_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!omap3beagle_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(omap3beagle_snd_device, &omap3beagle_snd_devdata);
+ omap3beagle_snd_devdata.dev = &omap3beagle_snd_device->dev;
+ *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(omap3beagle_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(omap3beagle_snd_device);
+
+ return ret;
+}
+
+static void __exit omap3beagle_soc_exit(void)
+{
+ platform_device_unregister(omap3beagle_snd_device);
+}
+
+module_init(omap3beagle_soc_init);
+module_exit(omap3beagle_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
new file mode 100644
index 00000000000..bd91594496b
--- /dev/null
+++ b/sound/soc/omap/omap3pandora.c
@@ -0,0 +1,311 @@
+/*
+ * omap3pandora.c -- SoC audio for Pandora Handheld Console
+ *
+ * Author: GraÅžvydas Ignotas <notasas@gmail.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/gpio.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define OMAP3_PANDORA_DAC_POWER_GPIO 118
+#define OMAP3_PANDORA_AMP_POWER_GPIO 14
+
+#define PREFIX "ASoC omap3pandora: "
+
+static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
+ struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set codec system clock\n");
+ return ret;
+ }
+
+ /* Set McBSP clock to external */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set cpu system clock\n");
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
+ if (ret < 0) {
+ pr_err(PREFIX "can't set SRG clock divider\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int omap3pandora_out_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;
+
+ return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_in_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;
+
+ return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
+ gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
+ } else {
+ gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+ mdelay(1);
+ gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+ }
+
+ return 0;
+}
+
+/*
+ * Audio paths on Pandora board:
+ *
+ * |O| ---> PCM DAC +-> AMP -> Headphone Jack
+ * |M| A +--------> Line Out
+ * |A| <~~clk~~+
+ * |P| <--- TWL4030 <--------- Line In and MICs
+ */
+static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
+ 0, 0, NULL, 0, omap3pandora_hp_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+};
+
+static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+ SND_SOC_DAPM_MIC("Mic (external)", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+ {"Headphone Amplifier", NULL, "PCM DAC"},
+ {"Line Out", NULL, "PCM DAC"},
+ {"Headphone Jack", NULL, "Headphone Amplifier"},
+};
+
+static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
+ {"INL", NULL, "Line In"},
+ {"INR", NULL, "Line In"},
+ {"INL", NULL, "Mic (Internal)"},
+ {"INR", NULL, "Mic (external)"},
+};
+
+static int omap3pandora_out_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
+ ARRAY_SIZE(omap3pandora_out_dapm_widgets));
+ if (ret < 0)
+ return ret;
+
+ snd_soc_dapm_add_routes(codec, omap3pandora_out_map,
+ ARRAY_SIZE(omap3pandora_out_map));
+
+ return snd_soc_dapm_sync(codec);
+}
+
+static int omap3pandora_in_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
+ ARRAY_SIZE(omap3pandora_in_dapm_widgets));
+ if (ret < 0)
+ return ret;
+
+ snd_soc_dapm_add_routes(codec, omap3pandora_in_map,
+ ARRAY_SIZE(omap3pandora_in_map));
+
+ return snd_soc_dapm_sync(codec);
+}
+
+static struct snd_soc_ops omap3pandora_out_ops = {
+ .hw_params = omap3pandora_out_hw_params,
+};
+
+static struct snd_soc_ops omap3pandora_in_ops = {
+ .hw_params = omap3pandora_in_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+ {
+ .name = "PCM1773",
+ .stream_name = "HiFi Out",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap3pandora_out_ops,
+ .init = omap3pandora_out_init,
+ }, {
+ .name = "TWL4030",
+ .stream_name = "Line/Mic In",
+ .cpu_dai = &omap_mcbsp_dai[1],
+ .codec_dai = &twl4030_dai,
+ .ops = &omap3pandora_in_ops,
+ .init = omap3pandora_in_init,
+ }
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_omap3pandora = {
+ .name = "omap3pandora",
+ .platform = &omap_soc_platform,
+ .dai_link = omap3pandora_dai,
+ .num_links = ARRAY_SIZE(omap3pandora_dai),
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3pandora_snd_data = {
+ .card = &snd_soc_card_omap3pandora,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3pandora_snd_device;
+
+static int __init omap3pandora_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap3_pandora()) {
+ pr_debug(PREFIX "Not OMAP3 Pandora\n");
+ return -ENODEV;
+ }
+ pr_info("OMAP3 Pandora SoC init\n");
+
+ ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
+ if (ret) {
+ pr_err(PREFIX "Failed to get DAC power GPIO\n");
+ return ret;
+ }
+
+ ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+ if (ret) {
+ pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
+ goto fail0;
+ }
+
+ ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
+ if (ret) {
+ pr_err(PREFIX "Failed to get amp power GPIO\n");
+ goto fail0;
+ }
+
+ ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+ if (ret) {
+ pr_err(PREFIX "Failed to set amp power GPIO direction\n");
+ goto fail1;
+ }
+
+ omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
+ if (omap3pandora_snd_device == NULL) {
+ pr_err(PREFIX "Platform device allocation failed\n");
+ ret = -ENOMEM;
+ goto fail1;
+ }
+
+ platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data);
+ omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev;
+ *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */
+ *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */
+
+ ret = platform_device_add(omap3pandora_snd_device);
+ if (ret) {
+ pr_err(PREFIX "Unable to add platform device\n");
+ goto fail2;
+ }
+
+ return 0;
+
+fail2:
+ platform_device_put(omap3pandora_snd_device);
+fail1:
+ gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+fail0:
+ gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+ return ret;
+}
+module_init(omap3pandora_soc_init);
+
+static void __exit omap3pandora_soc_exit(void)
+{
+ platform_device_unregister(omap3pandora_snd_device);
+ gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+ gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+}
+module_exit(omap3pandora_soc_exit);
+
+MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 00000000000..cd41a948df7
--- /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_B |
+ 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_B |
+ 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_card snd_soc_card_osk = {
+ .name = "OSK5912",
+ .platform = &omap_soc_platform,
+ .dai_link = &osk_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+ .card = &snd_soc_card_osk,
+ .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");
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
new file mode 100644
index 00000000000..a72dc4e159e
--- /dev/null
+++ b/sound/soc/omap/overo.c
@@ -0,0 +1,148 @@
+/*
+ * overo.c -- SoC audio for Gumstix Overo
+ *
+ * Author: Steve Sakoman <steve@sakoman.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 <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int overo_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 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops overo_ops = {
+ .hw_params = overo_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link overo_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &overo_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_overo = {
+ .name = "overo",
+ .platform = &omap_soc_platform,
+ .dai_link = &overo_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device overo_snd_devdata = {
+ .card = &snd_soc_card_overo,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *overo_snd_device;
+
+static int __init overo_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_overo()) {
+ pr_debug("Not Overo!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "overo SoC init\n");
+
+ overo_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!overo_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(overo_snd_device, &overo_snd_devdata);
+ overo_snd_devdata.dev = &overo_snd_device->dev;
+ *(unsigned int *)overo_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(overo_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(overo_snd_device);
+
+ return ret;
+}
+module_init(overo_soc_init);
+
+static void __exit overo_soc_exit(void)
+{
+ platform_device_unregister(overo_snd_device);
+}
+module_exit(overo_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC overo");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
new file mode 100644
index 00000000000..ad97836818b
--- /dev/null
+++ b/sound/soc/omap/sdp3430.c
@@ -0,0 +1,152 @@
+/*
+ * sdp3430.c -- SoC audio for TI OMAP3430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * Based on:
+ * Author: Steve Sakoman <steve@sakoman.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 <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int sdp3430_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 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ 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_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops sdp3430_ops = {
+ .hw_params = sdp3430_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp3430_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai,
+ .ops = &sdp3430_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+ .name = "SDP3430",
+ .platform = &omap_soc_platform,
+ .dai_link = &sdp3430_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device sdp3430_snd_devdata = {
+ .machine = &snd_soc_machine_sdp3430,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *sdp3430_snd_device;
+
+static int __init sdp3430_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap_3430sdp()) {
+ pr_debug("Not SDP3430!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "SDP3430 SoC init\n");
+
+ sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!sdp3430_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
+ sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
+ *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+ ret = platform_device_add(sdp3430_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(sdp3430_snd_device);
+
+ return ret;
+}
+module_init(sdp3430_soc_init);
+
+static void __exit sdp3430_soc_exit(void)
+{
+ platform_device_unregister(sdp3430_snd_device);
+}
+module_exit(sdp3430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP3430");
+MODULE_LICENSE("GPL");
+
generated by cgit v1.2.3 (git 2.25.1) at 2024-08-05 11:48:40 +0000