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-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile2
-rw-r--r--sound/soc/s3c24xx/Kconfig27
-rw-r--r--sound/soc/s3c24xx/Makefile9
-rw-r--r--sound/soc/s3c24xx/lm4857.h32
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c670
-rw-r--r--sound/soc/s3c24xx/s3c2443-ac97.c401
-rw-r--r--sound/soc/s3c24xx/s3c24xx-ac97.h25
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c4
-rw-r--r--sound/soc/s3c24xx/smdk2443_wm9710.c85
-rw-r--r--sound/soc/sh/Kconfig38
-rw-r--r--sound/soc/sh/Makefile14
-rw-r--r--sound/soc/sh/dma-sh7760.c354
-rw-r--r--sound/soc/sh/hac.c322
-rw-r--r--sound/soc/sh/sh7760-ac97.c92
-rw-r--r--sound/soc/sh/ssi.c400
16 files changed, 2473 insertions, 3 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 10cffc08718..97b25523317 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -27,6 +27,7 @@ config SND_SOC
source "sound/soc/at91/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/sh/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0ae2e49036f..30414037763 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
-obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 044a3712077..e97c68306a9 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,7 @@
config SND_S3C24XX_SOC
tristate "SoC Audio for the Samsung S3C24XX chips"
depends on ARCH_S3C2410 && SND_SOC
+ select SND_PCM
help
Say Y or M if you want to add support for codecs attached to
the S3C24XX AC97, I2S or SSP interface. You will also need
@@ -8,3 +9,29 @@ config SND_S3C24XX_SOC
config SND_S3C24XX_SOC_I2S
tristate
+
+config SND_S3C2443_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+config SND_S3C24XX_SOC_NEO1973_WM8753
+ tristate "SoC I2S Audio support for NEO1973 - WM8753"
+ depends on SND_S3C24XX_SOC && MACH_GTA01
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_WM8753
+ help
+ Say Y if you want to add support for SoC audio on smdk2440
+ with the WM8753.
+
+config SND_S3C24XX_SOC_SMDK2443_WM9710
+ tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
+ depends on SND_S3C24XX_SOC && MACH_SMDK2443
+ select SND_S3C2443_SOC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ Say Y if you want to add support for SoC audio on smdk2443
+ with the WM9710.
+
+
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 6f0fffcb30f..13c92f0fa1e 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -1,6 +1,15 @@
# S3c24XX Platform Support
snd-soc-s3c24xx-objs := s3c24xx-pcm.o
snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
+snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
+obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
+
+# S3C24XX Machine Support
+snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
+
+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h
new file mode 100644
index 00000000000..0cf5b7011d6
--- /dev/null
+++ b/sound/soc/s3c24xx/lm4857.h
@@ -0,0 +1,32 @@
+/*
+ * lm4857.h -- ALSA Soc Audio Layer
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * 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.
+ *
+ * Revision history
+ * 18th Jun 2007 Initial version.
+ */
+
+#ifndef LM4857_H_
+#define LM4857_H_
+
+/* The register offsets in the cache array */
+#define LM4857_MVOL 0
+#define LM4857_LVOL 1
+#define LM4857_RVOL 2
+#define LM4857_CTRL 3
+
+/* the shifts required to set these bits */
+#define LM4857_3D 5
+#define LM4857_WAKEUP 5
+#define LM4857_EPGAIN 4
+
+#endif /*LM4857_H_*/
+
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
new file mode 100644
index 00000000000..d5a8fc2cf8d
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -0,0 +1,670 @@
+/*
+ * neo1973_wm8753.c -- SoC audio for Neo1973
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * 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.
+ *
+ * Revision history
+ * 20th Jan 2007 Initial version.
+ * 05th Feb 2007 Rename all to Neo1973
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+#include <asm/arch/regs-iis.h>
+#include <asm/arch/regs-clock.h>
+#include <asm/arch/regs-gpio.h>
+#include <asm/hardware.h>
+#include <asm/arch/audio.h>
+#include <asm/io.h>
+#include <asm/arch/spi-gpio.h>
+#include "../codecs/wm8753.h"
+#include "lm4857.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+/* define the scenarios */
+#define NEO_AUDIO_OFF 0
+#define NEO_GSM_CALL_AUDIO_HANDSET 1
+#define NEO_GSM_CALL_AUDIO_HEADSET 2
+#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
+#define NEO_STEREO_TO_SPEAKERS 4
+#define NEO_STEREO_TO_HEADPHONES 5
+#define NEO_CAPTURE_HANDSET 6
+#define NEO_CAPTURE_HEADSET 7
+#define NEO_CAPTURE_BLUETOOTH 8
+
+static struct snd_soc_machine neo1973;
+static struct i2c_client *i2c;
+
+static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pll_out = 0, bclk = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ pll_out = 12288000;
+ break;
+ case 48000:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 12288000;
+ break;
+ case 96000:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 12288000;
+ break;
+ case 11025:
+ bclk = WM8753_BCLK_DIV_16;
+ pll_out = 11289600;
+ break;
+ case 22050:
+ bclk = WM8753_BCLK_DIV_8;
+ pll_out = 11289600;
+ break;
+ case 44100:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 11289600;
+ break;
+ case 88200:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set MCLK division for sample rate */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_32FS );
+ if (ret < 0)
+ return ret;
+
+ /* set codec BCLK division for sample rate */
+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(4,4));
+ if (ret < 0)
+ return ret;
+
+ /* codec PLL input is PCLK/4 */
+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1,
+ iis_clkrate / 4, pll_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * Neo1973 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops neo1973_hifi_ops = {
+ .hw_params = neo1973_hifi_hw_params,
+ .hw_free = neo1973_hifi_hw_free,
+};
+
+static int neo1973_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int pcmdiv = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ if (params_rate(params) != 8000)
+ return -EINVAL;
+ if (params_channels(params) != 1)
+ return -EINVAL;
+
+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
+ /* todo: gg check mode (DSP_B) against CSR datasheet */
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set codec PCM division for sample rate */
+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
+ if (ret < 0)
+ return ret;
+
+ /* configue and enable PLL for 12.288MHz output */
+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2,
+ iis_clkrate / 4, 12288000);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+
+ /* disable the PLL */
+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops neo1973_voice_ops = {
+ .hw_params = neo1973_voice_hw_params,
+ .hw_free = neo1973_voice_hw_free,
+};
+
+static int neo1973_scenario = 0;
+
+static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = neo1973_scenario;
+ return 0;
+}
+
+static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
+{
+ switch(neo1973_scenario) {
+ case NEO_AUDIO_OFF:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_GSM_CALL_AUDIO_HANDSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+ break;
+ case NEO_GSM_CALL_AUDIO_HEADSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_GSM_CALL_AUDIO_BLUETOOTH:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_STEREO_TO_SPEAKERS:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_STEREO_TO_HEADPHONES:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_CAPTURE_HANDSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
+ break;
+ case NEO_CAPTURE_HEADSET:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ case NEO_CAPTURE_BLUETOOTH:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
+ default:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+
+ return 0;
+}
+
+static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (neo1973_scenario == ucontrol->value.integer.value[0])
+ return 0;
+
+ neo1973_scenario = ucontrol->value.integer.value[0];
+ set_scenario_endpoints(codec, neo1973_scenario);
+ return 1;
+}
+
+static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
+
+static void lm4857_write_regs(void)
+{
+ if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
+ printk(KERN_ERR "lm4857: i2c write failed\n");
+}
+
+static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int reg=kcontrol->private_value & 0xFF;
+ int shift = (kcontrol->private_value >> 8) & 0x0F;
+ int mask = (kcontrol->private_value >> 16) & 0xFF;
+
+ ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
+ return 0;
+}
+
+static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int reg = kcontrol->private_value & 0xFF;
+ int shift = (kcontrol->private_value >> 8) & 0x0F;
+ int mask = (kcontrol->private_value >> 16) & 0xFF;
+
+ if (((lm4857_regs[reg] >> shift ) & mask) ==
+ ucontrol->value.integer.value[0])
+ return 0;
+
+ lm4857_regs[reg] &= ~ (mask << shift);
+ lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
+ lm4857_write_regs();
+ return 1;
+}
+
+static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
+
+ if (value)
+ value -= 5;
+
+ ucontrol->value.integer.value[0] = value;
+ return 0;
+}
+
+static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = ucontrol->value.integer.value[0];
+
+ if (value)
+ value += 5;
+
+ if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
+ return 0;
+
+ lm4857_regs[LM4857_CTRL] &= 0xF0;
+ lm4857_regs[LM4857_CTRL] |= value;
+ lm4857_write_regs();
+ return 1;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Audio Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Call Mic", NULL),
+};
+
+
+/* example machine audio_mapnections */
+static const char* audio_map[][3] = {
+
+ /* Connections to the lm4857 amp */
+ {"Audio Out", NULL, "LOUT1"},
+ {"Audio Out", NULL, "ROUT1"},
+
+ /* Connections to the GSM Module */
+ {"GSM Line Out", NULL, "MONO1"},
+ {"GSM Line Out", NULL, "MONO2"},
+ {"RXP", NULL, "GSM Line In"},
+ {"RXN", NULL, "GSM Line In"},
+
+ /* Connections to Headset */
+ {"MIC1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Mic"},
+
+ /* Call Mic */
+ {"MIC2", NULL, "Mic Bias"},
+ {"MIC2N", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Call Mic"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+
+ {NULL, NULL, NULL},
+};
+
+static const char *lm4857_mode[] = {
+ "Off",
+ "Call Speaker",
+ "Stereo Speakers",
+ "Stereo Speakers + Headphones",
+ "Headphones"
+};
+
+static const struct soc_enum lm4857_mode_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
+};
+
+static const char *neo_scenarios[] = {
+ "Off",
+ "GSM Handset",
+ "GSM Headset",
+ "GSM Bluetooth",
+ "Speakers",
+ "Headphones",
+ "Capture Handset",
+ "Capture Headset",
+ "Capture Bluetooth"
+};
+
+static const struct soc_enum neo_scenario_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios),
+};
+
+static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
+ SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
+ lm4857_get_mode, lm4857_set_mode),
+ SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
+ neo1973_get_scenario, neo1973_set_scenario),
+ SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 II. It is missing logic to detect hp/mic insertions and logic
+ * to re-route the audio in such an event.
+ */
+static int neo1973_wm8753_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* set up NC codec pins */
+ snd_soc_dapm_set_endpoint(codec, "LOUT2", 0);
+ snd_soc_dapm_set_endpoint(codec, "ROUT2", 0);
+ snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+ snd_soc_dapm_set_endpoint(codec, "OUT4", 0);
+ snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
+ snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
+
+
+ /* set endpoints to default mode */
+ set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
+ /* Add neo1973 specific widgets */
+ for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
+
+ /* add neo1973 specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8753_neo1973_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* set up neo1973 specific audio path audio_mapnects */
+ for (i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_cpu_dai bt_dai =
+{ .name = "Bluetooth",
+ .id = 0,
+ .type = SND_SOC_DAI_PCM,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+};
+
+static struct snd_soc_dai_link neo1973_dai[] = {
+{ /* Hifi Playback - for similatious use with voice below */
+ .name = "WM8753",
+ .stream_name = "WM8753 HiFi",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
+ .init = neo1973_wm8753_init,
+ .ops = &neo1973_hifi_ops,
+},
+{ /* Voice via BT */
+ .name = "Bluetooth",
+ .stream_name = "Voice",
+ .cpu_dai = &bt_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+ .ops = &neo1973_voice_ops,
+},
+};
+
+static struct snd_soc_machine neo1973 = {
+ .name = "neo1973",
+ .dai_link = neo1973_dai,
+ .num_links = ARRAY_SIZE(neo1973_dai),
+};
+
+static struct wm8753_setup_data neo1973_wm8753_setup = {
+ .i2c_address = 0x1a,
+};
+
+static struct snd_soc_device neo1973_snd_devdata = {
+ .machine = &neo1973,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8753,
+ .codec_data = &neo1973_wm8753_setup,
+};
+
+static struct i2c_client client_template;
+
+static unsigned short normal_i2c[] = { 0x7C, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ int ret;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL)
+ return -ENOMEM;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ printk(KERN_ERR "LM4857 failed to attach at addr %x\n", addr);
+ goto exit_err;
+ }
+
+ lm4857_write_regs();
+ return ret;
+
+exit_err:
+ kfree(i2c);
+ return ret;
+}
+
+static int lm4857_i2c_detach(struct i2c_client *client)
+{
+ i2c_detach_client(client);
+ kfree(client);
+ return 0;
+}
+
+static int lm4857_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, lm4857_amp_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver lm4857_i2c_driver = {
+ .driver = {
+ .name = "LM4857 I2C Amp",
+ .owner = THIS_MODULE,
+ },
+ .id = I2C_DRIVERID_LM4857,
+ .attach_adapter = lm4857_i2c_attach,
+ .detach_client = lm4857_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "LM4857",
+ .driver = &lm4857_i2c_driver,
+};
+
+static struct platform_device *neo1973_snd_device;
+
+static int __init neo1973_init(void)
+{
+ int ret;
+
+ neo1973_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!neo1973_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata);
+ neo1973_snd_devdata.dev = &neo1973_snd_device->dev;
+ ret = platform_device_add(neo1973_snd_device);
+
+ if (ret)
+ platform_device_put(neo1973_snd_device);
+
+ ret = i2c_add_driver(&lm4857_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+
+ return ret;
+}
+
+static void __exit neo1973_exit(void)
+{
+ platform_device_unregister(neo1973_snd_device);
+}
+
+module_init(neo1973_init);
+module_exit(neo1973_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
new file mode 100644
index 00000000000..75acf7ef552
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -0,0 +1,401 @@
+/*
+ * s3c2443-ac97.c -- ALSA Soc Audio Layer
+ *
+ * (c) 2007 Wolfson Microelectronics PLC.
+ * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * Copyright (C) 2005, Sean Choi <sh428.choi@samsung.com>
+ * All rights reserved.
+ *
+ * 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.
+ *
+ * Revision history
+ * 21st Mar 2007 Initial Version
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/hardware.h>
+#include <asm/io.h>
+#include <asm/arch/regs-ac97.h>
+#include <asm/arch/regs-gpio.h>
+#include <asm/arch/regs-clock.h>
+#include <asm/arch/audio.h>
+#include <asm/dma.h>
+#include <asm/arch/dma.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-ac97.h"
+
+struct s3c24xx_ac97_info {
+ void __iomem *regs;
+ struct clk *ac97_clk;
+};
+static struct s3c24xx_ac97_info s3c24xx_ac97;
+
+DECLARE_COMPLETION(ac97_completion);
+static u32 codec_ready;
+static DECLARE_MUTEX(ac97_mutex);
+
+static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ u32 ac_glbctrl;
+ u32 ac_codec_cmd;
+ u32 stat, addr, data;
+
+ down(&ac97_mutex);
+
+ codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
+ ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+ ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg);
+ writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+
+ udelay(50);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ wait_for_completion(&ac97_completion);
+
+ stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT);
+ addr = (stat >> 16) & 0x7f;
+ data = (stat & 0xffff);
+
+ if (addr != reg)
+ printk(KERN_ERR "s3c24xx-ac97: req addr = %02x,"
+ " rep addr = %02x\n", reg, addr);
+
+ up(&ac97_mutex);
+
+ return (unsigned short)data;
+}
+
+static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ u32 ac_glbctrl;
+ u32 ac_codec_cmd;
+
+ down(&ac97_mutex);
+
+ codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
+ ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+ ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val);
+ writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+
+ udelay(50);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ wait_for_completion(&ac97_completion);
+
+ ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+ ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ;
+ writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+
+ up(&ac97_mutex);
+
+}
+
+static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = 0;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+}
+
+static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = 0;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA |
+ S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+}
+
+static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id)
+{
+ int status;
+ u32 ac_glbctrl;
+
+ status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready;
+
+ if (status) {
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ complete(&ac97_completion);
+ }
+ return IRQ_HANDLED;
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = s3c2443_ac97_read,
+ .write = s3c2443_ac97_write,
+ .warm_reset = s3c2443_ac97_warm_reset,
+ .reset = s3c2443_ac97_cold_reset,
+};
+
+static struct s3c2410_dma_client s3c2443_dma_client_out = {
+ .name = "AC97 PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c2443_dma_client_in = {
+ .name = "AC97 PCM Stereo in"
+};
+
+static struct s3c2410_dma_client s3c2443_dma_client_micin = {
+ .name = "AC97 Mic Mono in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = {
+ .client = &s3c2443_dma_client_out,
+ .channel = DMACH_PCM_OUT,
+ .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
+ .dma_size = 4,
+};
+
+static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = {
+ .client = &s3c2443_dma_client_in,
+ .channel = DMACH_PCM_IN,
+ .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
+ .dma_size = 4,
+};
+
+static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = {
+ .client = &s3c2443_dma_client_micin,
+ .channel = DMACH_MIC_IN,
+ .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA,
+ .dma_size = 4,
+};
+
+static int s3c2443_ac97_probe(struct platform_device *pdev)
+{
+ int ret;
+ u32 ac_glbctrl;
+
+ s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100);
+ if (s3c24xx_ac97.regs == NULL)
+ return -ENXIO;
+
+ s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97");
+ if (s3c24xx_ac97.ac97_clk == NULL) {
+ printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n");
+ iounmap(s3c24xx_ac97.regs);
+ return -ENODEV;
+ }
+ clk_enable(s3c24xx_ac97.ac97_clk);
+
+ s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET);
+ s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC);
+ s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI);
+ s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = 0;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq,
+ IRQF_DISABLED, "AC97", NULL);
+ if (ret < 0) {
+ printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n");
+ clk_disable(s3c24xx_ac97.ac97_clk);
+ clk_put(s3c24xx_ac97.ac97_clk);
+ iounmap(s3c24xx_ac97.regs);
+ }
+ return ret;
+}
+
+static void s3c2443_ac97_remove(struct platform_device *pdev)
+{
+ free_irq(IRQ_S3C2443_AC97, NULL);
+ clk_disable(s3c24xx_ac97.ac97_clk);
+ clk_put(s3c24xx_ac97.ac97_clk);
+ iounmap(s3c24xx_ac97.regs);
+}
+
+static int s3c2443_ac97_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out;
+ else
+ cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in;
+
+ return 0;
+}
+
+static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ switch(cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA;
+ else
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK;
+ else
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK;
+ break;
+ }
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ return 0;
+}
+
+static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+ else
+ cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in;
+
+ return 0;
+}
+
+static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ switch(cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK;
+ }
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ return 0;
+}
+
+#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_cpu_dai s3c2443_ac97_dai[] = {
+{
+ .name = "s3c2443-ac97",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .probe = s3c2443_ac97_probe,
+ .remove = s3c2443_ac97_remove,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = s3c2443_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = s3c2443_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = s3c2443_ac97_hw_params,
+ .trigger = s3c2443_ac97_trigger},
+},
+{
+ .name = "pxa2xx-ac97-mic",
+ .id = 1,
+ .type = SND_SOC_DAI_AC97,
+ .capture = {
+ .stream_name = "AC97 Mic Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = s3c2443_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = s3c2443_ac97_hw_mic_params,
+ .trigger = s3c2443_ac97_mic_trigger,},
+},
+};
+
+EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+MODULE_AUTHOR("Graeme Gregory");
+MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
new file mode 100644
index 00000000000..2b835e8260f
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx-ac97.h
@@ -0,0 +1,25 @@
+/*
+ * s3c24xx-ac97.c -- ALSA Soc Audio Layer
+ *
+ * (c) 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * 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.
+ *
+ * Revision history
+ * 10th Nov 2006 Initial version.
+ */
+
+#ifndef S3C24XXAC97_H_
+#define S3C24XXAC97_H_
+
+#define AC_CMD_ADDR(x) (x << 16)
+#define AC_CMD_DATA(x) (x & 0xffff)
+
+extern struct snd_soc_cpu_dai s3c2443_ac97_dai[];
+
+#endif /*S3C24XXAC97_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 8ca314dc889..39f02462e07 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -344,11 +344,11 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
DBG("Entered %s\n", __FUNCTION__);
switch (div_id) {
- case S3C24XX_DIV_MCLK:
+ case S3C24XX_DIV_BCLK:
reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK;
writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
break;
- case S3C24XX_DIV_BCLK:
+ case S3C24XX_DIV_MCLK:
reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS);
writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
break;
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
new file mode 100644
index 00000000000..d46cd811ceb
--- /dev/null
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -0,0 +1,85 @@
+/*
+ * smdk2443_wm9710.c -- SoC audio for smdk2443
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * 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.
+ *
+ * Revision history
+ * 8th Mar 2007 Initial version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/ac97.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-ac97.h"
+
+static struct snd_soc_machine smdk2443;
+
+static struct snd_soc_dai_link smdk2443_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &s3c2443_ac97_dai[0],
+ .codec_dai = &ac97_dai,
+},
+};
+
+static struct snd_soc_machine smdk2443 = {
+ .name = "SMDK2443",
+ .dai_link = smdk2443_dai,
+ .num_links = ARRAY_SIZE(smdk2443_dai),
+};
+
+static struct snd_soc_device smdk2443_snd_ac97_devdata = {
+ .machine = &smdk2443,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+static struct platform_device *smdk2443_snd_ac97_device;
+
+static int __init smdk2443_init(void)
+{
+ int ret;
+
+ smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!smdk2443_snd_ac97_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(smdk2443_snd_ac97_device,
+ &smdk2443_snd_ac97_devdata);
+ smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev;
+ ret = platform_device_add(smdk2443_snd_ac97_device);
+
+ if (ret)
+ platform_device_put(smdk2443_snd_ac97_device);
+
+ return ret;
+}
+
+static void __exit smdk2443_exit(void)
+{
+ platform_device_unregister(smdk2443_snd_ac97_device);
+}
+
+module_init(smdk2443_init);
+module_exit(smdk2443_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
new file mode 100644
index 00000000000..f03220d23e7
--- /dev/null
+++ b/sound/soc/sh/Kconfig
@@ -0,0 +1,38 @@
+menu "SoC Audio support for SuperH"
+
+config SND_SOC_PCM_SH7760
+ tristate "SoC Audio support for Renesas SH7760"
+ depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG
+ help
+ Enable this option for SH7760 AC97/I2S audio support.
+
+
+##
+## Audio unit modules
+##
+
+config SND_SOC_SH4_HAC
+ select AC97_BUS
+ select SND_SOC_AC97_BUS
+ select SND_AC97_CODEC
+ tristate
+
+config SND_SOC_SH4_SSI
+ tristate
+
+
+
+##
+## Boards
+##
+
+config SND_SH7760_AC97
+ tristate "SH7760 AC97 sound support"
+ depends on CPU_SUBTYPE_SH7760 && SND_SOC_PCM_SH7760
+ select SND_SOC_SH4_HAC
+ select SND_SOC_AC97_CODEC
+ help
+ This option enables generic sound support for the first
+ AC97 unit of the SH7760.
+
+endmenu
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
new file mode 100644
index 00000000000..a8e8ab81cc6
--- /dev/null
+++ b/sound/soc/sh/Makefile
@@ -0,0 +1,14 @@
+## DMA engines
+snd-soc-dma-sh7760-objs := dma-sh7760.o
+obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o
+
+## audio units found on some SH-4
+snd-soc-hac-objs := hac.o
+snd-soc-ssi-objs := ssi.o
+obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o
+obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o
+
+## boards
+snd-soc-sh7760-ac97-objs := sh7760-ac97.o
+
+obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
new file mode 100644
index 00000000000..cdee374b843
--- /dev/null
+++ b/sound/soc/sh/dma-sh7760.c
@@ -0,0 +1,354 @@
+/*
+ * SH7760 ("camelot") DMABRG audio DMA unit support
+ *
+ * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ * licensed under the terms outlined in the file COPYING at the root
+ * of the linux kernel sources.
+ *
+ * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
+ * trigger an interrupt when one half of the programmed transfer size
+ * has been xmitted.
+ *
+ * FIXME: little-endian only for now
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/dmabrg.h>
+
+
+/* registers and bits */
+#define BRGATXSAR 0x00
+#define BRGARXDAR 0x04
+#define BRGATXTCR 0x08
+#define BRGARXTCR 0x0C
+#define BRGACR 0x10
+#define BRGATXTCNT 0x14
+#define BRGARXTCNT 0x18
+
+#define ACR_RAR (1 << 18)
+#define ACR_RDS (1 << 17)
+#define ACR_RDE (1 << 16)
+#define ACR_TAR (1 << 2)
+#define ACR_TDS (1 << 1)
+#define ACR_TDE (1 << 0)
+
+/* receiver/transmitter data alignment */
+#define ACR_RAM_NONE (0 << 24)
+#define ACR_RAM_4BYTE (1 << 24)
+#define ACR_RAM_2WORD (2 << 24)
+#define ACR_TAM_NONE (0 << 8)
+#define ACR_TAM_4BYTE (1 << 8)
+#define ACR_TAM_2WORD (2 << 8)
+
+
+struct camelot_pcm {
+ unsigned long mmio; /* DMABRG audio channel control reg MMIO */
+ unsigned int txid; /* ID of first DMABRG IRQ for this unit */
+
+ struct snd_pcm_substream *tx_ss;
+ unsigned long tx_period_size;
+ unsigned int tx_period;
+
+ struct snd_pcm_substream *rx_ss;
+ unsigned long rx_period_size;
+ unsigned int rx_period;
+
+} cam_pcm_data[2] = {
+ {
+ .mmio = 0xFE3C0040,
+ .txid = DMABRGIRQ_A0TXF,
+ },
+ {
+ .mmio = 0xFE3C0060,
+ .txid = DMABRGIRQ_A1TXF,
+ },
+};
+
+#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x)))
+
+/*
+ * set a minimum of 16kb per period, to avoid interrupt-"storm" and
+ * resulting skipping. In general, the bigger the minimum size, the
+ * better for overall system performance. (The SH7760 is a puny CPU
+ * with a slow SDRAM interface and poor internal bus bandwidth,
+ * *especially* when the LCDC is active). The minimum for the DMAC
+ * is 8 bytes; 16kbytes are enough to get skip-free playback of a
+ * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain
+ * reasonable responsiveness in MPlayer.
+ */
+#define DMABRG_PERIOD_MIN 16 * 1024
+#define DMABRG_PERIOD_MAX 0x03fffffc
+#define DMABRG_PREALLOC_BUFFER 32 * 1024
+#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024
+
+/* support everything the SSI supports */
+#define DMABRG_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+#define DMABRG_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
+
+static struct snd_pcm_hardware camelot_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = DMABRG_FMTS,
+ .rates = DMABRG_RATES,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 8, /* max of the SSI */
+ .buffer_bytes_max = DMABRG_PERIOD_MAX,
+ .period_bytes_min = DMABRG_PERIOD_MIN,
+ .period_bytes_max = DMABRG_PERIOD_MAX / 2,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 128,
+};
+
+static void camelot_txdma(void *data)
+{
+ struct camelot_pcm *cam = data;
+ cam->tx_period ^= 1;
+ snd_pcm_period_elapsed(cam->tx_ss);
+}
+
+static void camelot_rxdma(void *data)
+{
+ struct camelot_pcm *cam = data;
+ cam->rx_period ^= 1;
+ snd_pcm_period_elapsed(cam->rx_ss);
+}
+
+static int camelot_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ int ret, dmairq;
+
+ snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware);
+
+ /* DMABRG buffer half/full events */
+ dmairq = (recv) ? cam->txid + 2 : cam->txid;
+ if (recv) {
+ cam->rx_ss = substream;
+ ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam);
+ if (unlikely(ret)) {
+ pr_debug("audio unit %d irqs already taken!\n",
+ rtd->dai->cpu_dai->id);
+ return -EBUSY;
+ }
+ (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam);
+ } else {
+ cam->tx_ss = substream;
+ ret = dmabrg_request_irq(dmairq, camelot_txdma, cam);
+ if (unlikely(ret)) {
+ pr_debug("audio unit %d irqs already taken!\n",
+ rtd->dai->cpu_dai->id);
+ return -EBUSY;
+ }
+ (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam);
+ }
+ return 0;
+}
+
+static int camelot_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ int dmairq;
+
+ dmairq = (recv) ? cam->txid + 2 : cam->txid;
+
+ if (recv)
+ cam->rx_ss = NULL;
+ else
+ cam->tx_ss = NULL;
+
+ dmabrg_free_irq(dmairq + 1);
+ dmabrg_free_irq(dmairq);
+
+ return 0;
+}
+
+static int camelot_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ int ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (ret < 0)
+ return ret;
+
+ if (recv) {
+ cam->rx_period_size = params_period_bytes(hw_params);
+ cam->rx_period = 0;
+ } else {
+ cam->tx_period_size = params_period_bytes(hw_params);
+ cam->tx_period = 0;
+ }
+ return 0;
+}
+
+static int camelot_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int camelot_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+
+ pr_debug("PCM data: addr 0x%08ulx len %d\n",
+ (u32)runtime->dma_addr, runtime->dma_bytes);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area;
+ BRGREG(BRGATXTCR) = runtime->dma_bytes;
+ } else {
+ BRGREG(BRGARXDAR) = (unsigned long)runtime->dma_area;
+ BRGREG(BRGARXTCR) = runtime->dma_bytes;
+ }
+
+ return 0;
+}
+
+static inline void dmabrg_play_dma_start(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* start DMABRG engine: XFER start, auto-addr-reload */
+ BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD;
+}
+
+static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* forcibly terminate data transmission */
+ BRGREG(BRGACR) = acr | ACR_TDS;
+}
+
+static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* start DMABRG engine: recv start, auto-reload */
+ BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD;
+}
+
+static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* forcibly terminate data receiver */
+ BRGREG(BRGACR) = acr | ACR_RDS;
+}
+
+static int camelot_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (recv)
+ dmabrg_rec_dma_start(cam);
+ else
+ dmabrg_play_dma_start(cam);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (recv)
+ dmabrg_rec_dma_stop(cam);
+ else
+ dmabrg_play_dma_stop(cam);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ unsigned long pos;
+
+ /* cannot use the DMABRG pointer register: under load, by the
+ * time ALSA comes around to read the register, it is already
+ * far ahead (or worse, already done with the fragment) of the
+ * position at the time the IRQ was triggered, which results in
+ * fast-playback sound in my test application (ScummVM)
+ */
+ if (recv)
+ pos = cam->rx_period ? cam->rx_period_size : 0;
+ else
+ pos = cam->tx_period ? cam->tx_period_size : 0;
+
+ return bytes_to_frames(runtime, pos);
+}
+
+static struct snd_pcm_ops camelot_pcm_ops = {
+ .open = camelot_pcm_open,
+ .close = camelot_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = camelot_hw_params,
+ .hw_free = camelot_hw_free,
+ .prepare = camelot_prepare,
+ .trigger = camelot_trigger,
+ .pointer = camelot_pos,
+};
+
+static void camelot_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int camelot_pcm_new(struct snd_card *card,
+ struct snd_soc_codec_dai *dai,
+ struct snd_pcm *pcm)
+{
+ /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
+ * in MMAP mode (i.e. aplay -M)
+ */
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX);
+
+ return 0;
+}
+
+struct snd_soc_platform sh7760_soc_platform = {
+ .name = "sh7760-pcm",
+ .pcm_ops = &camelot_pcm_ops,
+ .pcm_new = camelot_pcm_new,
+ .pcm_free = camelot_pcm_free,
+};
+EXPORT_SYMBOL_GPL(sh7760_soc_platform);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
new file mode 100644
index 00000000000..8e3f03908cd
--- /dev/null
+++ b/sound/soc/sh/hac.c
@@ -0,0 +1,322 @@
+/*
+ * Hitachi Audio Controller (AC97) support for SH7760/SH7780
+ *
+ * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ * licensed under the terms outlined in the file COPYING at the root
+ * of the linux kernel sources.
+ *
+ * dont forget to set IPSEL/OMSEL register bits (in your board code) to
+ * enable HAC output pins!
+ */
+
+/* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only
+ * the FIRST can be used since ASoC does not pass any information to the
+ * ac97_read/write() functions regarding WHICH unit to use. You'll have
+ * to edit the code a bit to use the other AC97 unit. --mlau
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+/* regs and bits */
+#define HACCR 0x08
+#define HACCSAR 0x20
+#define HACCSDR 0x24
+#define HACPCML 0x28
+#define HACPCMR 0x2C
+#define HACTIER 0x50
+#define HACTSR 0x54
+#define HACRIER 0x58
+#define HACRSR 0x5C
+#define HACACR 0x60
+
+#define CR_CR (1 << 15) /* "codec-ready" indicator */
+#define CR_CDRT (1 << 11) /* cold reset */
+#define CR_WMRT (1 << 10) /* warm reset */
+#define CR_B9 (1 << 9) /* the mysterious "bit 9" */
+#define CR_ST (1 << 5) /* AC97 link start bit */
+
+#define CSAR_RD (1 << 19) /* AC97 data read bit */
+#define CSAR_WR (0)
+
+#define TSR_CMDAMT (1 << 31)
+#define TSR_CMDDMT (1 << 30)
+
+#define RSR_STARY (1 << 22)
+#define RSR_STDRY (1 << 21)
+
+#define ACR_DMARX16 (1 << 30)
+#define ACR_DMATX16 (1 << 29)
+#define ACR_TX12ATOM (1 << 26)
+#define ACR_DMARX20 ((1 << 24) | (1 << 22))
+#define ACR_DMATX20 ((1 << 23) | (1 << 21))
+
+#define CSDR_SHIFT 4
+#define CSDR_MASK (0xffff << CSDR_SHIFT)
+#define CSAR_SHIFT 12
+#define CSAR_MASK (0x7f << CSAR_SHIFT)
+
+#define AC97_WRITE_RETRY 1
+#define AC97_READ_RETRY 5
+
+/* manual-suggested AC97 codec access timeouts (us) */
+#define TMO_E1 500 /* 21 < E1 < 1000 */
+#define TMO_E2 13 /* 13 < E2 */
+#define TMO_E3 21 /* 21 < E3 */
+#define TMO_E4 500 /* 21 < E4 < 1000 */
+
+struct hac_priv {
+ unsigned long mmio; /* HAC base address */
+} hac_cpu_data[] = {
+#if defined(CONFIG_CPU_SUBTYPE_SH7760)
+ {
+ .mmio = 0xFE240000,
+ },
+ {
+ .mmio = 0xFE250000,
+ },
+#elif defined(CONFIG_CPU_SUBTYPE_SH7780)
+ {
+ .mmio = 0xFFE40000,
+ },
+#else
+#error "Unsupported SuperH SoC"
+#endif
+};
+
+#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg)))
+
+/*
+ * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906)
+ */
+static int hac_get_codec_data(struct hac_priv *hac, unsigned short r,
+ unsigned short *v)
+{
+ unsigned int to1, to2, i;
+ unsigned short adr;
+
+ for (i = 0; i < AC97_READ_RETRY; ++i) {
+ *v = 0;
+ /* wait for HAC to receive something from the codec */
+ for (to1 = TMO_E4;
+ to1 && !(HACREG(HACRSR) & RSR_STARY);
+ --to1)
+ udelay(1);
+ for (to2 = TMO_E4;
+ to2 && !(HACREG(HACRSR) & RSR_STDRY);
+ --to2)
+ udelay(1);
+
+ if (!to1 && !to2)
+ return 0; /* codec comm is down */
+
+ adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT);
+ *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT);
+
+ HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY);
+
+ if (r == adr)
+ break;
+
+ /* manual says: wait at least 21 usec before retrying */
+ udelay(21);
+ }
+ HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY);
+ return (i < AC97_READ_RETRY);
+}
+
+static unsigned short hac_read_codec_aux(struct hac_priv *hac,
+ unsigned short reg)
+{
+ unsigned short val;
+ unsigned int i, to;
+
+ for (i = 0; i < AC97_READ_RETRY; i++) {
+ /* send_read_request */
+ local_irq_disable();
+ HACREG(HACTSR) &= ~(TSR_CMDAMT);
+ HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD;
+ local_irq_enable();
+
+ for (to = TMO_E3;
+ to && !(HACREG(HACTSR) & TSR_CMDAMT);
+ --to)
+ udelay(1);
+
+ HACREG(HACTSR) &= ~TSR_CMDAMT;
+ val = 0;
+ if (hac_get_codec_data(hac, reg, &val) != 0)
+ break;
+ }
+
+ if (i == AC97_READ_RETRY)
+ return ~0;
+
+ return val;
+}
+
+static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac = &hac_cpu_data[unit_id];
+ unsigned int i, to;
+ /* write_codec_aux */
+ for (i = 0; i < AC97_WRITE_RETRY; i++) {
+ /* send_write_request */
+ local_irq_disable();
+ HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT);
+ HACREG(HACCSDR) = (val << CSDR_SHIFT);
+ HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD);
+ local_irq_enable();
+
+ /* poll-wait for CMDAMT and CMDDMT */
+ for (to = TMO_E1;
+ to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT));
+ --to)
+ udelay(1);
+
+ HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT);
+ if (to)
+ break;
+ /* timeout, try again */
+ }
+}
+
+static unsigned short hac_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac = &hac_cpu_data[unit_id];
+ return hac_read_codec_aux(hac, reg);
+}
+
+static void hac_ac97_warmrst(struct snd_ac97 *ac97)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac = &hac_cpu_data[unit_id];
+ unsigned int tmo;
+
+ HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9;
+ msleep(10);
+ HACREG(HACCR) = CR_ST | CR_B9;
+ for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--)
+ udelay(1);
+
+ if (!tmo)
+ printk(KERN_INFO "hac: reset: AC97 link down!\n");
+ /* settings this bit lets us have a conversation with codec */
+ HACREG(HACACR) |= ACR_TX12ATOM;
+}
+
+static void hac_ac97_coldrst(struct snd_ac97 *ac97)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac;
+ hac = &hac_cpu_data[unit_id];
+
+ HACREG(HACCR) = 0;
+ HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9;
+ msleep(10);
+ hac_ac97_warmrst(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = hac_ac97_read,
+ .write = hac_ac97_write,
+ .reset = hac_ac97_coldrst,
+ .warm_reset = hac_ac97_warmrst,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int hac_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
+ int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ switch (params->msbits) {
+ case 16:
+ HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16;
+ HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20;
+ break;
+ case 20:
+ HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16;
+ HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20;
+ break;
+ default:
+ pr_debug("hac: invalid depth %d bit\n", params->msbits);
+ return -EINVAL;
+ break;
+ }
+
+ return 0;
+}
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+#define AC97_FMTS \
+ SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_cpu_dai sh4_hac_dai[] = {
+{
+ .name = "HAC0",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = {
+ .hw_params = hac_hw_params,
+ },
+},
+#ifdef CONFIG_CPU_SUBTYPE_SH7760
+{
+ .name = "HAC1",
+ .id = 1,
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = {
+ .hw_params = hac_hw_params,
+ },
+
+},
+#endif
+};
+EXPORT_SYMBOL_GPL(sh4_hac_dai);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
new file mode 100644
index 00000000000..5563f14511f
--- /dev/null
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -0,0 +1,92 @@
+/*
+ * Generic AC97 sound support for SH7760
+ *
+ * (c) 2007 Manuel Lauss
+ *
+ * Licensed under the GPLv2.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/io.h>
+
+#include "../codecs/ac97.h"
+
+#define IPSEL 0xFE400034
+
+/* platform specific structs can be declared here */
+extern struct snd_soc_cpu_dai sh4_hac_dai[2];
+extern struct snd_soc_platform sh7760_soc_platform;
+
+static int machine_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
+static struct snd_soc_dai_link sh7760_ac97_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &sh4_hac_dai[0], /* HAC0 */
+ .codec_dai = &ac97_dai,
+ .init = machine_init,
+ .ops = NULL,
+};
+
+static struct snd_soc_machine sh7760_ac97_soc_machine = {
+ .name = "SH7760 AC97",
+ .dai_link = &sh7760_ac97_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device sh7760_ac97_snd_devdata = {
+ .machine = &sh7760_ac97_soc_machine,
+ .platform = &sh7760_soc_platform,
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+static struct platform_device *sh7760_ac97_snd_device;
+
+static int __init sh7760_ac97_init(void)
+{
+ int ret;
+ unsigned short ipsel;
+
+ /* enable both AC97 controllers in pinmux reg */
+ ipsel = ctrl_inw(IPSEL);
+ ctrl_outw(ipsel | (3 << 10), IPSEL);
+
+ ret = -ENOMEM;
+ sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!sh7760_ac97_snd_device)
+ goto out;
+
+ platform_set_drvdata(sh7760_ac97_snd_device,
+ &sh7760_ac97_snd_devdata);
+ sh7760_ac97_snd_devdata.dev = &sh7760_ac97_snd_device->dev;
+ ret = platform_device_add(sh7760_ac97_snd_device);
+
+ if (ret)
+ platform_device_put(sh7760_ac97_snd_device);
+
+out:
+ return ret;
+}
+
+static void __exit sh7760_ac97_exit(void)
+{
+ platform_device_unregister(sh7760_ac97_snd_device);
+}
+
+module_init(sh7760_ac97_init);
+module_exit(sh7760_ac97_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
new file mode 100644
index 00000000000..b72bc316cb8
--- /dev/null
+++ b/sound/soc/sh/ssi.c
@@ -0,0 +1,400 @@
+/*
+ * Serial Sound Interface (I2S) support for SH7760/SH7780
+ *
+ * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * licensed under the terms outlined in the file COPYING at the root
+ * of the linux kernel sources.
+ *
+ * dont forget to set IPSEL/OMSEL register bits (in your board code) to
+ * enable SSI output pins!
+ */
+
+/*
+ * LIMITATIONS:
+ * The SSI unit has only one physical data line, so full duplex is
+ * impossible. This can be remedied on the SH7760 by using the
+ * other SSI unit for recording; however the SH7780 has only 1 SSI
+ * unit, and its pins are shared with the AC97 unit, among others.
+ *
+ * FEATURES:
+ * The SSI features "compressed mode": in this mode it continuously
+ * streams PCM data over the I2S lines and uses LRCK as a handshake
+ * signal. Can be used to send compressed data (AC3/DTS) to a DSP.
+ * The number of bits sent over the wire in a frame can be adjusted
+ * and can be independent from the actual sample bit depth. This is
+ * useful to support TDM mode codecs like the AD1939 which have a
+ * fixed TDM slot size, regardless of sample resolution.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/io.h>
+
+#define SSICR 0x00
+#define SSISR 0x04
+
+#define CR_DMAEN (1 << 28)
+#define CR_CHNL_SHIFT 22
+#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT)
+#define CR_DWL_SHIFT 19
+#define CR_DWL_MASK (7 << CR_DWL_SHIFT)
+#define CR_SWL_SHIFT 16
+#define CR_SWL_MASK (7 << CR_SWL_SHIFT)
+#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */
+#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */
+#define CR_SCKP (1 << 13) /* I2Sclock polarity */
+#define CR_SWSP (1 << 12) /* LRCK polarity */
+#define CR_SPDP (1 << 11)
+#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */
+#define CR_PDTA (1 << 9) /* fifo data alignment */
+#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */
+#define CR_BREN (1 << 7) /* clock gating in burst mode */
+#define CR_CKDIV_SHIFT 4
+#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */
+#define CR_MUTE (1 << 3) /* SSI mute */
+#define CR_CPEN (1 << 2) /* compressed mode */
+#define CR_TRMD (1 << 1) /* transmit/receive select */
+#define CR_EN (1 << 0) /* enable SSI */
+
+#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg)))
+
+struct ssi_priv {
+ unsigned long mmio;
+ unsigned long sysclk;
+ int inuse;
+} ssi_cpu_data[] = {
+#if defined(CONFIG_CPU_SUBTYPE_SH7760)
+ {
+ .mmio = 0xFE680000,
+ },
+ {
+ .mmio = 0xFE690000,
+ },
+#elif defined(CONFIG_CPU_SUBTYPE_SH7780)
+ {
+ .mmio = 0xFFE70000,
+ },
+#else
+#error "Unsupported SuperH SoC"
+#endif
+};
+
+/*
+ * track usage of the SSI; it is simplex-only so prevent attempts of
+ * concurrent playback + capture. FIXME: any locking required?
+ */
+static int ssi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+ if (ssi->inuse) {
+ pr_debug("ssi: already in use!\n");
+ return -EBUSY;
+ } else
+ ssi->inuse = 1;
+ return 0;
+}
+
+static void ssi_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+
+ ssi->inuse = 0;
+}
+
+static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ SSIREG(SSICR) |= CR_DMAEN | CR_EN;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ssi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+ unsigned long ssicr = SSIREG(SSICR);
+ unsigned int bits, channels, swl, recv, i;
+
+ channels = params_channels(params);
+ bits = params->msbits;
+ recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1;
+
+ pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr);
+ pr_debug("bits: %d channels: %d\n", bits, channels);
+
+ ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA |
+ CR_SWL_MASK);
+
+ /* direction (send/receive) */
+ if (!recv)
+ ssicr |= CR_TRMD; /* transmit */
+
+ /* channels */
+ if ((channels < 2) || (channels > 8) || (channels & 1)) {
+ pr_debug("ssi: invalid number of channels\n");
+ return -EINVAL;
+ }
+ ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT;
+
+ /* DATA WORD LENGTH (DWL): databits in audio sample */
+ i = 0;
+ switch (bits) {
+ case 32: ++i;
+ case 24: ++i;
+ case 22: ++i;
+ case 20: ++i;
+ case 18: ++i;
+ case 16: ++i;
+ ssicr |= i << CR_DWL_SHIFT;
+ case 8: break;
+ default:
+ pr_debug("ssi: invalid sample width\n");
+ return -EINVAL;
+ }
+
+ /*
+ * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S
+ * wires. This is usually bits_per_sample x channels/2; i.e. in
+ * Stereo mode the SWL equals DWL. SWL can be bigger than the
+ * product of (channels_per_slot x samplebits), e.g. for codecs
+ * like the AD1939 which only accept 32bit wide TDM slots. For
+ * "standard" I2S operation we set SWL = chans / 2 * DWL here.
+ * Waiting for ASoC to get TDM support ;-)
+ */
+ if ((bits > 16) && (bits <= 24)) {
+ bits = 24; /* these are padded by the SSI */
+ /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */
+ }
+ i = 0;
+ swl = (bits * channels) / 2;
+ switch (swl) {
+ case 256: ++i;
+ case 128: ++i;
+ case 64: ++i;
+ case 48: ++i;
+ case 32: ++i;
+ case 16: ++i;
+ ssicr |= i << CR_SWL_SHIFT;
+ case 8: break;
+ default:
+ pr_debug("ssi: invalid system word length computed\n");
+ return -EINVAL;
+ }
+
+ SSIREG(SSICR) = ssicr;
+
+ pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr);
+ return 0;
+}
+
+static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id];
+
+ ssi->sysclk = freq;
+
+ return 0;
+}
+
+/*
+ * This divider is used to generate the SSI_SCK (I2S bitclock) from the
+ * clock at the HAC_BIT_CLK ("oversampling clock") pin.
+ */
+static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div)
+{
+ struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
+ unsigned long ssicr;
+ int i;
+
+ i = 0;
+ ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK;
+ switch (div) {
+ case 16: ++i;
+ case 8: ++i;
+ case 4: ++i;
+ case 2: ++i;
+ SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT);
+ case 1: break;
+ default:
+ pr_debug("ssi: invalid sck divider %d\n", div);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt)
+{
+ struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
+ unsigned long ssicr = SSIREG(SSICR);
+
+ pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr);
+
+ ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP |
+ CR_SWS_MASTER | CR_SCK_MASTER);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ssicr |= CR_DEL | CR_PDTA;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ssicr |= CR_DEL;
+ break;
+ default:
+ pr_debug("ssi: unsupported format\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+ case SND_SOC_DAIFMT_CONT:
+ break;
+ case SND_SOC_DAIFMT_GATED:
+ ssicr |= CR_BREN;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ ssicr |= CR_SCKP; /* sample data at low clkedge */
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ssicr |= CR_SCKP | CR_SWSP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ ssicr |= CR_SWSP; /* word select starts low */
+ break;
+ default:
+ pr_debug("ssi: invalid inversion\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ ssicr |= CR_SCK_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ ssicr |= CR_SWS_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ssicr |= CR_SWS_MASTER | CR_SCK_MASTER;
+ break;
+ default:
+ pr_debug("ssi: invalid master/slave configuration\n");
+ return -EINVAL;
+ }
+
+ SSIREG(SSICR) = ssicr;
+ pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr);
+
+ return 0;
+}
+
+/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in
+ * Master mode, so really this is board specific; the SSI can do any
+ * rate with the right bitclk and divider settings.
+ */
+#define SSI_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+/* the SSI can do 8-32 bit samples, with 8 possible channels */
+#define SSI_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
+
+struct snd_soc_cpu_dai sh4_ssi_dai[] = {
+{
+ .name = "SSI0",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .playback = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .capture = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .ops = {
+ .startup = ssi_startup,
+ .shutdown = ssi_shutdown,
+ .trigger = ssi_trigger,
+ .hw_params = ssi_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = ssi_set_sysclk,
+ .set_clkdiv = ssi_set_clkdiv,
+ .set_fmt = ssi_set_fmt,
+ },
+},
+#ifdef CONFIG_CPU_SUBTYPE_SH7760
+{
+ .name = "SSI1",
+ .id = 1,
+ .type = SND_SOC_DAI_I2S,
+ .playback = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .capture = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .ops = {
+ .startup = ssi_startup,
+ .shutdown = ssi_shutdown,
+ .trigger = ssi_trigger,
+ .hw_params = ssi_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = ssi_set_sysclk,
+ .set_clkdiv = ssi_set_clkdiv,
+ .set_fmt = ssi_set_fmt,
+ },
+},
+#endif
+};
+EXPORT_SYMBOL_GPL(sh4_ssi_dai);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-02 07:11:07 +0000