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-rw-r--r--sound/soc/pxa/Kconfig42
-rw-r--r--sound/soc/pxa/Makefile8
-rw-r--r--sound/soc/pxa/brownstone.c174
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c33
-rw-r--r--sound/soc/pxa/mmp-pcm.c297
-rw-r--r--sound/soc/pxa/mmp-sspa.c480
-rw-r--r--sound/soc/pxa/mmp-sspa.h92
-rw-r--r--sound/soc/pxa/ttc-dkb.c173
8 files changed, 1278 insertions, 21 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index a0f7d3cfa47..4d2e46fae77 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -8,6 +8,15 @@ config SND_PXA2XX_SOC
the PXA2xx AC97, I2S or SSP interface. You will also need
to select the audio interfaces to support below.
+config SND_MMP_SOC
+ bool "Soc Audio for Marvell MMP chips"
+ depends on ARCH_MMP
+ select SND_SOC_DMAENGINE_PCM
+ select SND_ARM
+ help
+ Say Y if you want to add support for codecs attached to
+ the MMP SSPA interface.
+
config SND_PXA2XX_AC97
tristate
select SND_AC97_CODEC
@@ -26,6 +35,9 @@ config SND_PXA_SOC_SSP
tristate
select PXA_SSP
+config SND_MMP_SOC_SSPA
+ tristate
+
config SND_PXA2XX_SOC_CORGI
tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -138,6 +150,26 @@ config SND_SOC_TAVOREVB3
Say Y if you want to add support for SoC audio on the
Marvell Saarb reference platform.
+config SND_PXA910_SOC
+ tristate "SoC Audio for Marvell PXA910 chip"
+ depends on ARCH_MMP && SND
+ select SND_PCM
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell PXA910 reference platform.
+
+config SND_SOC_TTC_DKB
+ bool "SoC Audio support for TTC DKB"
+ depends on SND_PXA910_SOC && MACH_TTC_DKB
+ select PXA_SSP
+ select SND_PXA_SOC_SSP
+ select SND_MMP_SOC
+ select MFD_88PM860X
+ select SND_SOC_88PM860X
+ help
+ Say Y if you want to add support for SoC audio on TTC DKB
+
+
config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
depends on SND_PXA2XX_SOC && MACH_ZYLONITE
@@ -194,3 +226,13 @@ config SND_PXA2XX_SOC_IMOTE2
help
Say Y if you want to add support for SoC audio on the
IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+ tristate "SoC Audio support for Marvell Brownstone"
+ depends on SND_MMP_SOC && MACH_BROWNSTONE
+ select SND_MMP_SOC_SSPA
+ select MFD_WM8994
+ select SND_SOC_WM8994
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index af357623be9..d8a265d2d5d 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -3,11 +3,15 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o
snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
# PXA Machine Support
snd-soc-corgi-objs := corgi.o
@@ -28,6 +32,8 @@ snd-soc-mioa701-objs := mioa701_wm9713.o
snd-soc-z2-objs := z2.o
snd-soc-imote2-objs := imote2.o
snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -47,3 +53,5 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
obj-$(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
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
new file mode 100644
index 00000000000..5e666e03d33
--- /dev/null
+++ b/sound/soc/pxa/brownstone.c
@@ -0,0 +1,174 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+ {"Ext Spk", NULL, "SPKOUTLP"},
+ {"Ext Spk", NULL, "SPKOUTLN"},
+ {"Ext Spk", NULL, "SPKOUTRP"},
+ {"Ext Spk", NULL, "SPKOUTRN"},
+
+ {"Headset Stereophone", NULL, "HPOUT1L"},
+ {"Headset Stereophone", NULL, "HPOUT1R"},
+
+ {"IN1RN", NULL, "Headset Mic"},
+
+ {"DMIC1DAT", NULL, "MICBIAS1"},
+ {"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Main Mic");
+
+ /* set endpoints to not connected */
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "IN1LN");
+ snd_soc_dapm_nc_pin(dapm, "IN1LP");
+ snd_soc_dapm_nc_pin(dapm, "IN1RP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+ snd_soc_dapm_sync(dapm);
+
+ return 0;
+}
+
+static int brownstone_wm8994_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_dai *cpu_dai = rtd->cpu_dai;
+ int freq_out, sspa_mclk, sysclk;
+ int sspa_div;
+
+ if (params_rate(params) > 11025) {
+ freq_out = params_rate(params) * 512;
+ sysclk = params_rate(params) * 256;
+ sspa_mclk = params_rate(params) * 64;
+ } else {
+ freq_out = params_rate(params) * 1024;
+ sysclk = params_rate(params) * 512;
+ sspa_mclk = params_rate(params) * 64;
+ }
+ sspa_div = freq_out;
+ do_div(sspa_div, sspa_mclk);
+
+ snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+ /* set wm8994 sysclk */
+ snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+ return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+ .hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+ .name = "WM8994",
+ .stream_name = "WM8994 HiFi",
+ .cpu_dai_name = "mmp-sspa-dai.0",
+ .codec_dai_name = "wm8994-aif1",
+ .platform_name = "mmp-pcm-audio",
+ .codec_name = "wm8994-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &brownstone_ops,
+ .init = brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+ .name = "brownstone",
+ .dai_link = brownstone_wm8994_dai,
+ .num_links = ARRAY_SIZE(brownstone_wm8994_dai),
+
+ .controls = brownstone_dapm_control,
+ .num_controls = ARRAY_SIZE(brownstone_dapm_control),
+ .dapm_widgets = brownstone_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+ .dapm_routes = brownstone_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int __devinit brownstone_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ brownstone.dev = &pdev->dev;
+ ret = snd_soc_register_card(&brownstone);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int __devexit brownstone_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&brownstone);
+ return 0;
+}
+
+static struct platform_driver mmp_driver = {
+ .driver = {
+ .name = "brownstone-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = brownstone_probe,
+ .remove = __devexit_p(brownstone_remove),
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 9c585af59b5..8687c1c65d2 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -186,36 +186,27 @@ static struct snd_soc_card mioa701 = {
.num_links = ARRAY_SIZE(mioa701_dai),
};
-static struct platform_device *mioa701_snd_device;
-
-static int mioa701_wm9713_probe(struct platform_device *pdev)
+static int __devinit mioa701_wm9713_probe(struct platform_device *pdev)
{
- int ret;
+ int rc;
if (!machine_is_mioa701())
return -ENODEV;
- dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
- "lead to overheating and possible destruction of your device."
- "Do not use without a good knowledge of mio's board design!\n");
-
- mioa701_snd_device = platform_device_alloc("soc-audio", -1);
- if (!mioa701_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(mioa701_snd_device, &mioa701);
-
- ret = platform_device_add(mioa701_snd_device);
- if (!ret)
- return 0;
-
- platform_device_put(mioa701_snd_device);
- return ret;
+ mioa701.dev = &pdev->dev;
+ rc = snd_soc_register_card(&mioa701);
+ if (!rc)
+ dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+ "lead to overheating and possible destruction of your device."
+ " Do not use without a good knowledge of mio's board design!\n");
+ return rc;
}
static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
{
- platform_device_unregister(mioa701_snd_device);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
return 0;
}
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 00000000000..73ac5463c9e
--- /dev/null
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,297 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/mmp_audio.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/sram.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+ int ssp_id;
+ struct resource *dma_res;
+};
+
+#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \
+ SNDRV_PCM_INFO_MMAP_VALID | \
+ SNDRV_PCM_INFO_INTERLEAVED | \
+ SNDRV_PCM_INFO_PAUSE | \
+ SNDRV_PCM_INFO_RESUME)
+
+#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+ {
+ .info = MMP_PCM_INFO,
+ .formats = MMP_PCM_FORMATS,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+ {
+ .info = MMP_PCM_INFO,
+ .formats = MMP_PCM_FORMATS,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct pxa2xx_pcm_dma_params *dma_params;
+ struct dma_slave_config slave_config;
+ int ret;
+
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dma_params)
+ return 0;
+
+ ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config);
+ if (ret)
+ return ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ slave_config.dst_addr = dma_params->dev_addr;
+ slave_config.dst_maxburst = 4;
+ } else {
+ slave_config.src_addr = dma_params->dev_addr;
+ slave_config.src_maxburst = 4;
+ }
+
+ ret = dmaengine_slave_config(chan, &slave_config);
+ if (ret)
+ return ret;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct mmp_dma_data *dma_data = param;
+ bool found = false;
+ char *devname;
+
+ devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+ dma_data->ssp_id);
+ if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+ (chan->chan_id == dma_data->dma_res->start)) {
+ found = true;
+ }
+
+ kfree(devname);
+ return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct mmp_dma_data *dma_data;
+ struct resource *r;
+ int ret;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+ if (!r)
+ return -EBUSY;
+
+ snd_soc_set_runtime_hwparams(substream,
+ &mmp_pcm_hardware[substream->stream]);
+ dma_data = devm_kzalloc(&pdev->dev,
+ sizeof(struct mmp_dma_data), GFP_KERNEL);
+ if (dma_data == NULL)
+ return -ENOMEM;
+
+ dma_data->dma_res = r;
+ dma_data->ssp_id = cpu_dai->id;
+
+ ret = snd_dmaengine_pcm_open(substream, filter, dma_data);
+ if (ret) {
+ devm_kfree(&pdev->dev, dma_data);
+ return ret;
+ }
+
+ snd_dmaengine_pcm_set_data(substream, dma_data);
+ return 0;
+}
+
+static int mmp_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+
+ snd_dmaengine_pcm_close(substream);
+ devm_kfree(&pdev->dev, dma_data);
+ return 0;
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long off = vma->vm_pgoff;
+
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ return remap_pfn_range(vma, vma->vm_start,
+ __phys_to_pfn(runtime->dma_addr) + off,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops mmp_pcm_ops = {
+ .open = mmp_pcm_open,
+ .close = mmp_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = mmp_pcm_hw_params,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+ .mmap = mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+ struct gen_pool *gpool;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return;
+
+ for (stream = 0; stream < 2; stream++) {
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ gen_pool_free(gpool, (unsigned long)buf->area, size);
+ buf->area = NULL;
+ }
+
+ return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+ int stream)
+{
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+ struct gen_pool *gpool;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = substream->pcm->card->dev;
+ buf->private_data = NULL;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return -ENOMEM;
+
+ buf->area = (unsigned char *)gen_pool_alloc(gpool, size);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area);
+ buf->bytes = size;
+ return 0;
+}
+
+int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0, stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+
+ ret = mmp_pcm_preallocate_dma_buffer(substream, stream);
+ if (ret)
+ goto err;
+ }
+
+ return 0;
+
+err:
+ mmp_pcm_free_dma_buffers(pcm);
+ return ret;
+}
+
+struct snd_soc_platform_driver mmp_soc_platform = {
+ .ops = &mmp_pcm_ops,
+ .pcm_new = mmp_pcm_new,
+ .pcm_free = mmp_pcm_free_dma_buffers,
+};
+
+static __devinit int mmp_pcm_probe(struct platform_device *pdev)
+{
+ struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+ if (pdata) {
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+ pdata->buffer_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+ pdata->period_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+ pdata->buffer_max_capture;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+ pdata->period_max_capture;
+ }
+ return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int __devexit mmp_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+ .driver = {
+ .name = "mmp-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = mmp_pcm_probe,
+ .remove = __devexit_p(mmp_pcm_remove),
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 00000000000..4d6cb8a30fc
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,480 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+ struct ssp_device *sspa;
+ struct pxa2xx_pcm_dma_params *dma_params;
+ struct clk *audio_clk;
+ struct clk *sysclk;
+ int dai_fmt;
+ int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+ __raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+ return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_enable(priv->sysclk);
+ clk_enable(priv->sspa->clk);
+
+ return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable(priv->sspa->clk);
+ clk_disable(priv->sysclk);
+
+ return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (clk_id) {
+ case MMP_SSPA_CLK_AUDIO:
+ ret = clk_set_rate(priv->audio_clk, freq);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK_PLL:
+ case MMP_SSPA_CLK_VCXO:
+ /* not support yet */
+ return -EINVAL;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (pll_id) {
+ case MMP_SYSCLK:
+ ret = clk_set_rate(priv->sysclk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK:
+ ret = clk_set_rate(priv->sspa->clk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ u32 sspa_sp, sspa_ctrl;
+
+ /* check if we need to change anything at all */
+ if (sspa_priv->dai_fmt == fmt)
+ return 0;
+
+ /* we can only change the settings if the port is not in use */
+ if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+ (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+ dev_err(&sspa->pdev->dev,
+ "can't change hardware dai format: stream is in use\n");
+ return -EINVAL;
+ }
+
+ /* reset port settings */
+ sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+ sspa_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ sspa_sp |= SSPA_SP_MSL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspa_sp |= SSPA_SP_FSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sspa_sp |= SSPA_TXSP_FPER(63);
+ sspa_sp |= SSPA_SP_FWID(31);
+ sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ /*
+ * FIXME: hw issue, for the tx serial port,
+ * can not config the master/slave mode;
+ * so must clean this bit.
+ * The master/slave mode has been set in the
+ * rx port.
+ */
+ sspa_sp &= ~SSPA_SP_MSL;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+ /* Since we are configuring the timings for the format by hand
+ * we have to defer some things until hw_params() where we
+ * know parameters like the sample size.
+ */
+ sspa_priv->dai_fmt = fmt;
+ return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_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 *cpu_dai = rtd->cpu_dai;
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ struct pxa2xx_pcm_dma_params *dma_params;
+ u32 sspa_ctrl;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+ else
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+ sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+ sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+ sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+ } else {
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+ }
+
+ dma_params = &sspa_priv->dma_params[substream->stream];
+ dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ (sspa->phys_base + SSPA_TXD) :
+ (sspa->phys_base + SSPA_RXD);
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+ return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /*
+ * whatever playback or capture, must enable rx.
+ * this is a hw issue, so need check if rx has been
+ * enabled or not; if has been enabled by another
+ * stream, do not enable again.
+ */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_enable(sspa);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_enable(sspa);
+
+ sspa_priv->running_cnt++;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sspa_priv->running_cnt--;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_disable(sspa);
+
+ /* have no capture stream, disable rx port */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_disable(sspa);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+ snd_soc_dai_set_drvdata(dai, priv);
+ return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+ .startup = mmp_sspa_startup,
+ .shutdown = mmp_sspa_shutdown,
+ .trigger = mmp_sspa_trigger,
+ .hw_params = mmp_sspa_hw_params,
+ .set_sysclk = mmp_sspa_set_dai_sysclk,
+ .set_pll = mmp_sspa_set_dai_pll,
+ .set_fmt = mmp_sspa_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver mmp_sspa_dai = {
+ .probe = mmp_sspa_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 128,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .ops = &mmp_sspa_dai_ops,
+};
+
+static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+ struct sspa_priv *priv;
+ struct resource *res;
+
+ priv = devm_kzalloc(&pdev->dev,
+ sizeof(struct sspa_priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->sspa = devm_kzalloc(&pdev->dev,
+ sizeof(struct ssp_device), GFP_KERNEL);
+ if (priv->sspa == NULL)
+ return -ENOMEM;
+
+ priv->dma_params = devm_kzalloc(&pdev->dev,
+ 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL);
+ if (priv->dma_params == NULL)
+ return -ENOMEM;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (res == NULL)
+ return -ENOMEM;
+
+ priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res);
+ if (priv->sspa->mmio_base == NULL)
+ return -ENODEV;
+
+ priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(priv->sspa->clk))
+ return PTR_ERR(priv->sspa->clk);
+
+ priv->audio_clk = clk_get(NULL, "mmp-audio");
+ if (IS_ERR(priv->audio_clk))
+ return PTR_ERR(priv->audio_clk);
+
+ priv->sysclk = clk_get(NULL, "mmp-sysclk");
+ if (IS_ERR(priv->sysclk)) {
+ clk_put(priv->audio_clk);
+ return PTR_ERR(priv->sysclk);
+ }
+ clk_enable(priv->audio_clk);
+ priv->dai_fmt = (unsigned int) -1;
+ platform_set_drvdata(pdev, priv);
+
+ return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai);
+}
+
+static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+ struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+ clk_disable(priv->audio_clk);
+ clk_put(priv->audio_clk);
+ clk_put(priv->sysclk);
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+ .driver = {
+ .name = "mmp-sspa-dai",
+ .owner = THIS_MODULE,
+ },
+ .probe = asoc_mmp_sspa_probe,
+ .remove = __devexit_p(asoc_mmp_sspa_remove),
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 00000000000..ea365cb9e78
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD (0x00)
+#define SSPA_RXID (0x04)
+#define SSPA_RXCTL (0x08)
+#define SSPA_RXSP (0x0c)
+#define SSPA_RXFIFO_UL (0x10)
+#define SSPA_RXINT_MASK (0x14)
+#define SSPA_RXC (0x18)
+#define SSPA_RXFIFO_NOFS (0x1c)
+#define SSPA_RXFIFO_SIZE (0x20)
+
+#define SSPA_TXD (0x80)
+#define SSPA_TXID (0x84)
+#define SSPA_TXCTL (0x88)
+#define SSPA_TXSP (0x8c)
+#define SSPA_TXFIFO_LL (0x90)
+#define SSPA_TXINT_MASK (0x94)
+#define SSPA_TXC (0x98)
+#define SSPA_TXFIFO_NOFS (0x9c)
+#define SSPA_TXFIFO_SIZE (0xa0)
+
+/* SSPA Control Register */
+#define SSPA_CTL_XPH (1 << 31) /* Read Phase */
+#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */
+#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */
+#define SSPA_CTL_XFRLEN2_MASK (7 << 24)
+#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */
+#define SSPA_CTL_XWDLEN2_MASK (7 << 21)
+#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */
+#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */
+#define SSPA_CTL_XSSZ2_MASK (7 << 16)
+#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */
+#define SSPA_CTL_XFRLEN1_MASK (7 << 8)
+#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */
+#define SSPA_CTL_XWDLEN1_MASK (7 << 5)
+#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */
+#define SSPA_CTL_XSSZ1_MASK (7 << 0)
+#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */
+
+#define SSPA_CTL_8_BITS (0x0) /* Sample Size */
+#define SSPA_CTL_12_BITS (0x1)
+#define SSPA_CTL_16_BITS (0x2)
+#define SSPA_CTL_20_BITS (0x3)
+#define SSPA_CTL_24_BITS (0x4)
+#define SSPA_CTL_32_BITS (0x5)
+
+/* SSPA Serial Port Register */
+#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */
+#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */
+#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */
+#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */
+#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */
+#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */
+#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */
+#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */
+#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL 0
+#define MMP_SSPA_CLK_VCXO 1
+#define MMP_SSPA_CLK_AUDIO 3
+
+/* sspa pll id */
+#define MMP_SYSCLK 0
+#define MMP_SSPA_CLK 1
+
+#endif /* _MMP_SSPA_H */
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 00000000000..935491a8a77
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,173 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+ { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+ SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+ {"Headset Stereophone", NULL, "HS1"},
+ {"Headset Stereophone", NULL, "HS2"},
+
+ {"Ext Speaker", NULL, "LSP"},
+ {"Ext Speaker", NULL, "LSN"},
+
+ {"Lineout Out 1", NULL, "LINEOUT1"},
+ {"Lineout Out 2", NULL, "LINEOUT2"},
+
+ {"MIC1P", NULL, "Mic1 Bias"},
+ {"MIC1N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Ext Mic 1"},
+
+ {"MIC2P", NULL, "Mic1 Bias"},
+ {"MIC2N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Headset Mic 2"},
+
+ {"MIC3P", NULL, "Mic3 Bias"},
+ {"MIC3N", NULL, "Mic3 Bias"},
+ {"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ /* connected pins */
+ snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+ snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+ snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+ /* Headset jack detection */
+ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+ | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+ &hs_jack);
+ snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+ hs_jack_pins);
+ snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+ &mic_jack);
+ snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+ mic_jack_pins);
+
+ /* headphone, microphone detection & headset short detection */
+ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+ SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+ pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+ return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+ .name = "88pm860x i2s",
+ .stream_name = "audio playback",
+ .codec_name = "88pm860x-codec",
+ .platform_name = "mmp-pcm-audio",
+ .cpu_dai_name = "pxa-ssp-dai.1",
+ .codec_dai_name = "88pm860x-i2s",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+ .name = "ttc-dkb-hifi",
+ .dai_link = ttc_pm860x_hifi_dai,
+ .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+ .dapm_widgets = ttc_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+ .dapm_routes = ttc_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int __devinit ttc_dkb_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &ttc_dkb_card;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ 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 ttc_dkb_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 ttc_dkb_driver = {
+ .driver = {
+ .name = "ttc-dkb-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = ttc_dkb_probe,
+ .remove = __devexit_p(ttc_dkb_remove),
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");