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-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt1
-rw-r--r--include/sound/mpu401.h1
-rw-r--r--sound/drivers/Kconfig2
-rw-r--r--sound/drivers/mpu401/mpu401_uart.c2
-rw-r--r--sound/pci/hda/patch_realtek.c86
-rw-r--r--sound/pci/ice1712/ice1724.c3
-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile2
-rw-r--r--sound/soc/codecs/wm9712.c62
-rw-r--r--sound/soc/omap/Kconfig19
-rw-r--r--sound/soc/omap/Makefile11
-rw-r--r--sound/soc/omap/n810.c336
-rw-r--r--sound/soc/omap/omap-mcbsp.c414
-rw-r--r--sound/soc/omap/omap-mcbsp.h49
-rw-r--r--sound/soc/omap/omap-pcm.c357
-rw-r--r--sound/soc/omap/omap-pcm.h35
16 files changed, 1344 insertions, 37 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index fd4c32a031c9..0bbee38acd26 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -795,6 +795,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
lg-lw LG LW20/LW25 laptop
tcl TCL S700
clevo Clevo laptops (m520G, m665n)
+ medion Medion Rim 2150
test for testing/debugging purpose, almost all controls can be
adjusted. Appearing only when compiled with
$CONFIG_SND_DEBUG=y
diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h
index 68b634b75068..1f1d53f8830b 100644
--- a/include/sound/mpu401.h
+++ b/include/sound/mpu401.h
@@ -50,6 +50,7 @@
#define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */
#define MPU401_INFO_MMIO (1 << 3) /* MMIO access */
#define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */
+#define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */
#define MPU401_MODE_BIT_INPUT 0
#define MPU401_MODE_BIT_OUTPUT 1
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index fe85af1c5693..a78a8d045175 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -8,6 +8,8 @@ config SND_PCSP
tristate "Internal PC speaker support"
depends on X86_PC && HIGH_RES_TIMERS
depends on INPUT
+ depends on SND
+ select SND_PCM
help
If you don't have a sound card in your computer, you can include a
driver for the PC speaker which allows it to act like a primitive
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 18cca2457d44..2af09996a3d0 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -243,7 +243,7 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd,
#endif
}
mpu->write(mpu, cmd, MPU401C(mpu));
- if (ack) {
+ if (ack && !(mpu->info_flags & MPU401_INFO_NO_ACK)) {
ok = 0;
timeout = 10000;
while (!ok && timeout-- > 0) {
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cdda64b02f46..d9783a4263e0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -60,6 +60,7 @@ enum {
ALC880_TCL_S700,
ALC880_LG,
ALC880_LG_LW,
+ ALC880_MEDION_RIM,
#ifdef CONFIG_SND_DEBUG
ALC880_TEST,
#endif
@@ -2275,6 +2276,75 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
alc880_lg_lw_automute(codec);
}
+static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_input_mux alc880_medion_rim_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ },
+};
+
+static struct hda_verb alc880_medion_rim_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Internal Speaker */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_medion_rim_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ if (present)
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
+ else
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
+}
+
+static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == ALC880_HP_EVENT)
+ alc880_medion_rim_automute(codec);
+}
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
static struct hda_amp_list alc880_loopbacks[] = {
{ 0x0b, HDA_INPUT, 0 },
@@ -2882,6 +2952,7 @@ static const char *alc880_models[ALC880_MODEL_LAST] = {
[ALC880_F1734] = "F1734",
[ALC880_LG] = "lg",
[ALC880_LG_LW] = "lg-lw",
+ [ALC880_MEDION_RIM] = "medion",
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = "test",
#endif
@@ -2933,6 +3004,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
+ SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
@@ -3227,6 +3299,20 @@ static struct alc_config_preset alc880_presets[] = {
.unsol_event = alc880_lg_lw_unsol_event,
.init_hook = alc880_lg_lw_automute,
},
+ [ALC880_MEDION_RIM] = {
+ .mixers = { alc880_medion_rim_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_medion_rim_init_verbs,
+ alc_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_medion_rim_capture_source,
+ .unsol_event = alc880_medion_rim_unsol_event,
+ .init_hook = alc880_medion_rim_automute,
+ },
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = {
.mixers = { alc880_test_mixer },
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 4490422fb930..67350901772c 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2429,6 +2429,7 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
ICEREG1724(ice, MPU_CTRL),
(MPU401_INFO_INTEGRATED |
+ MPU401_INFO_NO_ACK |
MPU401_INFO_TX_IRQ),
ice->irq, 0,
&ice->rmidi[0])) < 0) {
@@ -2442,12 +2443,10 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
outb(inb(ICEREG1724(ice, IRQMASK)) &
~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX),
ICEREG1724(ice, IRQMASK));
-#if 0 /* for testing */
/* set watermarks */
outb(VT1724_MPU_RX_FIFO | 0x1,
ICEREG1724(ice, MPU_FIFO_WM));
outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
-#endif
}
}
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index a3b51df2bea1..18f28ac4bfe8 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -30,6 +30,7 @@ source "sound/soc/s3c24xx/Kconfig"
source "sound/soc/sh/Kconfig"
source "sound/soc/fsl/Kconfig"
source "sound/soc/davinci/Kconfig"
+source "sound/soc/omap/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index e489dbdde458..782db2127108 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-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index d2d79e182a45..76c1e2d33e7d 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -37,23 +37,23 @@ static int ac97_write(struct snd_soc_codec *codec,
* WM9712 register cache
*/
static const u16 wm9712_reg[] = {
- 0x6174, 0x8000, 0x8000, 0x8000, // 6
- 0x0f0f, 0xaaa0, 0xc008, 0x6808, // e
- 0xe808, 0xaaa0, 0xad00, 0x8000, // 16
- 0xe808, 0x3000, 0x8000, 0x0000, // 1e
- 0x0000, 0x0000, 0x0000, 0x000f, // 26
- 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
- 0x0000, 0xbb80, 0x0000, 0x0000, // 36
- 0x0000, 0x2000, 0x0000, 0x0000, // 3e
- 0x0000, 0x0000, 0x0000, 0x0000, // 46
- 0x0000, 0x0000, 0xf83e, 0xffff, // 4e
- 0x0000, 0x0000, 0x0000, 0xf83e, // 56
- 0x0008, 0x0000, 0x0000, 0x0000, // 5e
- 0xb032, 0x3e00, 0x0000, 0x0000, // 66
- 0x0000, 0x0000, 0x0000, 0x0000, // 6e
- 0x0000, 0x0000, 0x0000, 0x0006, // 76
- 0x0001, 0x0000, 0x574d, 0x4c12, // 7e
- 0x0000, 0x0000 // virtual hp mixers
+ 0x6174, 0x8000, 0x8000, 0x8000, /* 6 */
+ 0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */
+ 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */
+ 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */
+ 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+ 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */
+ 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+ 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */
+ 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */
+ 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */
+ 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */
+ 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+ 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
+ 0x0000, 0x0000 /* virtual hp mixers */
};
/* virtual HP mixers regs */
@@ -94,7 +94,7 @@ static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
-SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
@@ -165,7 +165,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec)
for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
+ snd_soc_cnew(&wm9712_snd_ac97_controls[i],
+ codec, NULL));
if (err < 0)
return err;
}
@@ -363,7 +364,6 @@ static const char *audio_map[][3] = {
{"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
{"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
{"Left HP Mixer", NULL, "ALC Sidetone Mux"},
- //{"Right HP Mixer", NULL, "HP Mixer"},
/* Right HP mixer */
{"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
@@ -454,15 +454,13 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec)
{
int i;
- for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
+ for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
- }
- /* set up audio path audio_mapnects */
- for(i = 0; audio_map[i][0] != NULL; i++) {
+ /* set up audio path connects */
+ 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]);
- }
+ audio_map[i][1], audio_map[i][2]);
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -540,7 +538,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
}
#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
struct snd_soc_codec_dai wm9712_dai[] = {
{
@@ -577,8 +576,6 @@ EXPORT_SYMBOL_GPL(wm9712_dai);
static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
{
- u16 reg;
-
switch (event) {
case SNDRV_CTL_POWER_D0: /* full On */
case SNDRV_CTL_POWER_D1: /* partial On */
@@ -633,7 +630,7 @@ static int wm9712_soc_resume(struct platform_device *pdev)
u16 *cache = codec->reg_cache;
ret = wm9712_reset(codec, 1);
- if (ret < 0){
+ if (ret < 0) {
printk(KERN_ERR "could not reset AC97 codec\n");
return ret;
}
@@ -642,9 +639,9 @@ static int wm9712_soc_resume(struct platform_device *pdev)
if (ret == 0) {
/* Sync reg_cache with the hardware after cold reset */
- for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
+ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) {
if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
- (i > 0x58 && i != 0x5c))
+ (i > 0x58 && i != 0x5c))
continue;
soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
}
@@ -757,7 +754,6 @@ struct snd_soc_codec_device soc_codec_dev_wm9712 = {
.suspend = wm9712_soc_suspend,
.resume = wm9712_soc_resume,
};
-
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
new file mode 100644
index 000000000000..0230d83e8e5e
--- /dev/null
+++ b/sound/soc/omap/Kconfig
@@ -0,0 +1,19 @@
+menu "SoC Audio for the Texas Instruments OMAP"
+
+config SND_OMAP_SOC
+ tristate "SoC Audio for the Texas Instruments OMAP chips"
+ depends on ARCH_OMAP && SND_SOC
+
+config SND_OMAP_SOC_MCBSP
+ tristate
+ select OMAP_MCBSP
+
+config SND_OMAP_SOC_N810
+ tristate "SoC Audio support for Nokia N810"
+ depends on SND_OMAP_SOC && MACH_NOKIA_N810
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TLV320AIC3X
+ help
+ Say Y if you want to add support for SoC audio on Nokia N810.
+
+endmenu
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
new file mode 100644
index 000000000000..d8d8d58075e3
--- /dev/null
+++ b/sound/soc/omap/Makefile
@@ -0,0 +1,11 @@
+# OMAP Platform Support
+snd-soc-omap-objs := omap-pcm.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+
+obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
+obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+
+# OMAP Machine Support
+snd-soc-n810-objs := n810.o
+
+obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
new file mode 100644
index 000000000000..83b1eb4e40f3
--- /dev/null
+++ b/sound/soc/omap/n810.c
@@ -0,0 +1,336 @@
+/*
+ * n810.c -- SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/gpio.h>
+#include <asm/arch/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic3x.h"
+
+#define RX44_HEADSET_AMP_GPIO 10
+#define RX44_SPEAKER_AMP_GPIO 101
+
+static struct clk *sys_clkout2;
+static struct clk *sys_clkout2_src;
+static struct clk *func96m_clk;
+
+static int n810_spk_func;
+static int n810_jack_func;
+
+static void n810_ext_control(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func);
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func);
+
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int n810_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ n810_ext_control(codec);
+ return clk_enable(sys_clkout2);
+}
+
+static void n810_shutdown(struct snd_pcm_substream *substream)
+{
+ clk_disable(sys_clkout2);
+}
+
+static int n810_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;
+ int err;
+
+ /* Set codec DAI configuration */
+ err = codec_dai->dai_ops.set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0)
+ return err;
+
+ /* Set cpu DAI configuration */
+ err = cpu_dai->dai_ops.set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0)
+ return err;
+
+ /* Set the codec system clock for DAC and ADC */
+ err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000,
+ SND_SOC_CLOCK_IN);
+
+ return err;
+}
+
+static struct snd_soc_ops n810_ops = {
+ .startup = n810_startup,
+ .hw_params = n810_hw_params,
+ .shutdown = n810_shutdown,
+};
+
+static int n810_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_spk_func;
+
+ return 0;
+}
+
+static int n810_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_spk_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_jack_func;
+
+ return 0;
+}
+
+static int n810_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_jack_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1);
+ else
+ omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0);
+
+ return 0;
+}
+
+static int n810_jack_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1);
+ else
+ omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
+ SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
+};
+
+static const char *audio_map[][3] = {
+ {"Headphone Jack", NULL, "HPLOUT"},
+ {"Headphone Jack", NULL, "HPROUT"},
+
+ {"Ext Spk", NULL, "LLOUT"},
+ {"Ext Spk", NULL, "RLOUT"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *jack_function[] = {"Off", "Headphone"};
+static const struct soc_enum n810_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(3, jack_function),
+};
+
+static const struct snd_kcontrol_new aic33_n810_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", n810_enum[0],
+ n810_get_spk, n810_set_spk),
+ SOC_ENUM_EXT("Jack Function", n810_enum[1],
+ n810_get_jack, n810_set_jack),
+};
+
+static int n810_aic33_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* Not connected */
+ snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
+ snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
+ snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+
+ /* Add N810 specific controls */
+ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add N810 specific widgets */
+ for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]);
+
+ /* Set up N810 specific audio path audio_map */
+ for (i = 0; i < ARRAY_SIZE(audio_map); 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;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link n810_dai = {
+ .name = "TLV320AIC33",
+ .stream_name = "AIC33",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &aic3x_dai,
+ .init = n810_aic33_init,
+ .ops = &n810_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_n810 = {
+ .name = "N810",
+ .dai_link = &n810_dai,
+ .num_links = 1,
+};
+
+/* Audio private data */
+static struct aic3x_setup_data n810_aic33_setup = {
+ .i2c_address = 0x18,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device n810_snd_devdata = {
+ .machine = &snd_soc_machine_n810,
+ .platform = &omap_soc_platform,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &n810_aic33_setup,
+};
+
+static struct platform_device *n810_snd_device;
+
+static int __init n810_soc_init(void)
+{
+ int err;
+ struct device *dev;
+
+ if (!machine_is_nokia_n810())
+ return -ENODEV;
+
+ n810_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!n810_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(n810_snd_device, &n810_snd_devdata);
+ n810_snd_devdata.dev = &n810_snd_device->dev;
+ *(unsigned int *)n810_dai.cpu_dai->private_data = 1; /* McBSP2 */
+ err = platform_device_add(n810_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &n810_snd_device->dev;
+
+ sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
+ if (IS_ERR(sys_clkout2_src)) {
+ dev_err(dev, "Could not get sys_clkout2_src clock\n");
+ return -ENODEV;
+ }
+ sys_clkout2 = clk_get(dev, "sys_clkout2");
+ if (IS_ERR(sys_clkout2)) {
+ dev_err(dev, "Could not get sys_clkout2\n");
+ goto err1;
+ }
+ /*
+ * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
+ * 96 MHz as its parent in order to get 12 MHz
+ */
+ func96m_clk = clk_get(dev, "func_96m_ck");
+ if (IS_ERR(func96m_clk)) {
+ dev_err(dev, "Could not get func 96M clock\n");
+ goto err2;
+ }
+ clk_set_parent(sys_clkout2_src, func96m_clk);
+ clk_set_rate(sys_clkout2, 12000000);
+
+ if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0)
+ BUG();
+ if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0)
+ BUG();
+ omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0);
+ omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0);
+
+ return 0;
+err2:
+ clk_put(sys_clkout2);
+ platform_device_del(n810_snd_device);
+err1:
+ platform_device_put(n810_snd_device);
+
+ return err;
+
+}
+
+static void __exit n810_soc_exit(void)
+{
+ platform_device_unregister(n810_snd_device);
+}
+
+module_init(n810_soc_init);
+module_exit(n810_soc_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("ALSA SoC Nokia N810");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
new file mode 100644
index 000000000000..40d87e6d0de8
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -0,0 +1,414 @@
+/*
+ * omap-mcbsp.c -- OMAP ALSA SoC DAI driver using McBSP port
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/arch/control.h>
+#include <asm/arch/dma.h>
+#include <asm/arch/mcbsp.h>
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_KNOT)
+
+struct omap_mcbsp_data {
+ unsigned int bus_id;
+ struct omap_mcbsp_reg_cfg regs;
+ /*
+ * Flags indicating is the bus already activated and configured by
+ * another substream
+ */
+ int active;
+ int configured;
+};
+
+#define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id)
+
+static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
+
+/*
+ * Stream DMA parameters. DMA request line and port address are set runtime
+ * since they are different between OMAP1 and later OMAPs
+ */
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
+{
+ { .name = "I2S PCM Stereo out", },
+ { .name = "I2S PCM Stereo in", },
+},
+};
+
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+static const int omap1_dma_reqs[][2] = {
+ { OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX },
+ { OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX },
+ { OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX },
+};
+static const unsigned long omap1_mcbsp_port[][2] = {
+ { OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
+ { OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
+ { OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 },
+};
+#else
+static const int omap1_dma_reqs[][2] = {};
+static const unsigned long omap1_mcbsp_port[][2] = {};
+#endif
+#if defined(CONFIG_ARCH_OMAP2420)
+static const int omap2420_dma_reqs[][2] = {
+ { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
+ { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
+};
+static const unsigned long omap2420_mcbsp_port[][2] = {
+ { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
+ { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
+};
+#else
+static const int omap2420_dma_reqs[][2] = {};
+static const unsigned long omap2420_mcbsp_port[][2] = {};
+#endif
+
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int err = 0;
+
+ if (!cpu_dai->active)
+ err = omap_mcbsp_request(mcbsp_data->bus_id);
+
+ return err;
+}
+
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+
+ if (!cpu_dai->active) {
+ omap_mcbsp_free(mcbsp_data->bus_id);
+ mcbsp_data->configured = 0;
+ }
+}
+
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!mcbsp_data->active++)
+ omap_mcbsp_start(mcbsp_data->bus_id);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (!--mcbsp_data->active)
+ omap_mcbsp_stop(mcbsp_data->bus_id);
+ break;
+ default:
+ err = -EINVAL;
+ }
+
+ return err;
+}
+
+static int omap_mcbsp_dai_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;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+ int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+ unsigned long port;
+
+ if (cpu_class_is_omap1()) {
+ dma = omap1_dma_reqs[bus_id][substream->stream];
+ port = omap1_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap2420()) {
+ dma = omap2420_dma_reqs[bus_id][substream->stream];
+ port = omap2420_mcbsp_port[bus_id][substream->stream];
+ } else {
+ /*
+ * TODO: Add support for 2430 and 3430
+ */
+ return -ENODEV;
+ }
+ omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
+ omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
+ cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+ if (mcbsp_data->configured) {
+ /* McBSP already configured by another stream */
+ return 0;
+ }
+
+ switch (params_channels(params)) {
+ case 2:
+ /* Set 1 word per (McBPSP) frame and use dual-phase frames */
+ regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE;
+ regs->rcr1 |= RFRLEN1(1 - 1);
+ regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE;
+ regs->xcr1 |= XFRLEN1(1 - 1);
+ break;
+ default:
+ /* Unsupported number of channels */
+ return -EINVAL;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ /* Set word lengths */
+ regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16);
+ regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16);
+ regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16);
+ regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16);
+ /* Set FS period and length in terms of bit clock periods */
+ regs->srgr2 |= FPER(16 * 2 - 1);
+ regs->srgr1 |= FWID(16 - 1);
+ break;
+ default:
+ /* Unsupported PCM format */
+ return -EINVAL;
+ }
+
+ omap_mcbsp_config(bus_id, &mcbsp_data->regs);
+ mcbsp_data->configured = 1;
+
+ return 0;
+}
+
+/*
+ * This must be called before _set_clkdiv and _set_sysclk since McBSP register
+ * cache is initialized here
+ */
+static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+ if (mcbsp_data->configured)
+ return 0;
+
+ memset(regs, 0, sizeof(*regs));
+ /* Generic McBSP register settings */
+ regs->spcr2 |= XINTM(3) | FREE;
+ regs->spcr1 |= RINTM(3);
+ regs->rcr2 |= RFIG;
+ regs->xcr2 |= XFIG;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* 1-bit data delay */
+ regs->rcr2 |= RDATDLY(1);
+ regs->xcr2 |= XDATDLY(1);
+ break;
+ default:
+ /* Unsupported data format */
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ /* McBSP master. Set FS and bit clocks as outputs */
+ regs->pcr0 |= FSXM | FSRM |
+ CLKXM | CLKRM;
+ /* Sample rate generator drives the FS */
+ regs->srgr2 |= FSGM;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* McBSP slave */
+ break;
+ default:
+ /* Unsupported master/slave configuration */
+ return -EINVAL;
+ }
+
+ /* Set bit clock (CLKX/CLKR) and FS polarities */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ /*
+ * Normal BCLK + FS.
+ * FS active low. TX data driven on falling edge of bit clock
+ * and RX data sampled on rising edge of bit clock.
+ */
+ regs->pcr0 |= FSXP | FSRP |
+ CLKXP | CLKRP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ regs->pcr0 |= CLKXP | CLKRP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ regs->pcr0 |= FSXP | FSRP;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+ if (div_id != OMAP_MCBSP_CLKGDV)
+ return -ENODEV;
+
+ regs->srgr1 |= CLKGDV(div - 1);
+
+ return 0;
+}
+
+static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
+ int clk_id)
+{
+ int sel_bit;
+ u16 reg;
+
+ if (cpu_class_is_omap1()) {
+ /* OMAP1's can use only external source clock */
+ if (unlikely(clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK))
+ return -EINVAL;
+ else
+ return 0;
+ }
+
+ switch (mcbsp_data->bus_id) {
+ case 0:
+ reg = OMAP2_CONTROL_DEVCONF0;
+ sel_bit = 2;
+ break;
+ case 1:
+ reg = OMAP2_CONTROL_DEVCONF0;
+ sel_bit = 6;
+ break;
+ /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
+ default:
+ return -EINVAL;
+ }
+
+ if (cpu_class_is_omap2()) {
+ if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
+ omap_ctrl_writel(omap_ctrl_readl(reg) &
+ ~(1 << sel_bit), reg);
+ } else {
+ omap_ctrl_writel(omap_ctrl_readl(reg) |
+ (1 << sel_bit), reg);
+ }
+ }
+
+ return 0;
+}
+
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+ int clk_id, unsigned int freq,
+ int dir)
+{
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+ int err = 0;
+
+ switch (clk_id) {
+ case OMAP_MCBSP_SYSCLK_CLK:
+ regs->srgr2 |= CLKSM;
+ break;
+ case OMAP_MCBSP_SYSCLK_CLKS_FCLK:
+ case OMAP_MCBSP_SYSCLK_CLKS_EXT:
+ err = omap_mcbsp_dai_set_clks_src(mcbsp_data, clk_id);
+ break;
+
+ case OMAP_MCBSP_SYSCLK_CLKX_EXT:
+ regs->srgr2 |= CLKSM;
+ case OMAP_MCBSP_SYSCLK_CLKR_EXT:
+ regs->pcr0 |= SCLKME;
+ break;
+ default:
+ err = -ENODEV;
+ }
+
+ return err;
+}
+
+struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = {
+{
+ .name = "omap-mcbsp-dai",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = OMAP_MCBSP_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = OMAP_MCBSP_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .startup = omap_mcbsp_dai_startup,
+ .shutdown = omap_mcbsp_dai_shutdown,
+ .trigger = omap_mcbsp_dai_trigger,
+ .hw_params = omap_mcbsp_dai_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = omap_mcbsp_dai_set_dai_fmt,
+ .set_clkdiv = omap_mcbsp_dai_set_clkdiv,
+ .set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
+ },
+ .private_data = &mcbsp_data[0].bus_id,
+},
+};
+EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("OMAP I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
new file mode 100644
index 000000000000..9965fd4b0427
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -0,0 +1,49 @@
+/*
+ * omap-mcbsp.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_I2S_H__
+#define __OMAP_I2S_H__
+
+/* Source clocks for McBSP sample rate generator */
+enum omap_mcbsp_clksrg_clk {
+ OMAP_MCBSP_SYSCLK_CLKS_FCLK, /* Internal FCLK */
+ OMAP_MCBSP_SYSCLK_CLKS_EXT, /* External CLKS pin */
+ OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
+ OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
+ OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
+};
+
+/* McBSP dividers */
+enum omap_mcbsp_div {
+ OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */
+};
+
+/*
+ * REVISIT: Preparation for the ASoC v2. Let the number of available links to
+ * be same than number of McBSP ports found in OMAP(s) we are compiling for.
+ */
+#define NUM_LINKS 1
+
+extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS];
+
+#endif
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
new file mode 100644
index 000000000000..62370202c649
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.c
@@ -0,0 +1,357 @@
+/*
+ * omap-pcm.c -- ALSA PCM interface for the OMAP SoC
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/arch/dma.h>
+#include "omap-pcm.h"
+
+static const struct snd_pcm_hardware omap_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 64 * 1024,
+ .periods_min = 2,
+ .periods_max = 255,
+ .buffer_bytes_max = 128 * 1024,
+};
+
+struct omap_runtime_data {
+ spinlock_t lock;
+ struct omap_pcm_dma_data *dma_data;
+ int dma_ch;
+ int period_index;
+};
+
+static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
+{
+ struct snd_pcm_substream *substream = data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ unsigned long flags;
+
+ if (cpu_is_omap1510()) {
+ /*
+ * OMAP1510 doesn't support DMA chaining so have to restart
+ * the transfer after all periods are transferred
+ */
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (prtd->period_index >= 0) {
+ if (++prtd->period_index == runtime->periods) {
+ prtd->period_index = 0;
+ omap_start_dma(prtd->dma_ch);
+ }
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ }
+
+ snd_pcm_period_elapsed(substream);
+}
+
+/* this may get called several times by oss emulation */
+static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+ int err = 0;
+
+ if (!dma_data)
+ return -ENODEV;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ if (prtd->dma_data)
+ return 0;
+ prtd->dma_data = dma_data;
+ err = omap_request_dma(dma_data->dma_req, dma_data->name,
+ omap_pcm_dma_irq, substream, &prtd->dma_ch);
+ if (!cpu_is_omap1510()) {
+ /*
+ * Link channel with itself so DMA doesn't need any
+ * reprogramming while looping the buffer
+ */
+ omap_dma_link_lch(prtd->dma_ch, prtd->dma_ch);
+ }
+
+ return err;
+}
+
+static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+
+ if (prtd->dma_data == NULL)
+ return 0;
+
+ if (!cpu_is_omap1510())
+ omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
+ omap_free_dma(prtd->dma_ch);
+ prtd->dma_data = NULL;
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ return 0;
+}
+
+static int omap_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ struct omap_pcm_dma_data *dma_data = prtd->dma_data;
+ struct omap_dma_channel_params dma_params;
+
+ memset(&dma_params, 0, sizeof(dma_params));
+ /*
+ * Note: Regardless of interface data formats supported by OMAP McBSP
+ * or EAC blocks, internal representation is always fixed 16-bit/sample
+ */
+ dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
+ dma_params.trigger = dma_data->dma_req;
+ dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
+ dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
+ dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC;
+ dma_params.src_start = runtime->dma_addr;
+ dma_params.dst_start = dma_data->port_addr;
+ } else {
+ dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT;
+ dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
+ dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
+ dma_params.src_start = dma_data->port_addr;
+ dma_params.dst_start = runtime->dma_addr;
+ }
+ /*
+ * Set DMA transfer frame size equal to ALSA period size and frame
+ * count as no. of ALSA periods. Then with DMA frame interrupt enabled,
+ * we can transfer the whole ALSA buffer with single DMA transfer but
+ * still can get an interrupt at each period bounary
+ */
+ dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2;
+ dma_params.frame_count = runtime->periods;
+ omap_set_dma_params(prtd->dma_ch, &dma_params);
+
+ omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+
+ return 0;
+}
+
+static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ int ret = 0;
+
+ spin_lock_irq(&prtd->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ prtd->period_index = 0;
+ omap_start_dma(prtd->dma_ch);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ prtd->period_index = -1;
+ omap_stop_dma(prtd->dma_ch);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ spin_unlock_irq(&prtd->lock);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ dma_addr_t ptr;
+ snd_pcm_uframes_t offset;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ptr = omap_get_dma_src_pos(prtd->dma_ch);
+ else
+ ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+
+ offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+
+ return offset;
+}
+
+static int omap_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
+
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd = kzalloc(sizeof(prtd), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ spin_lock_init(&prtd->lock);
+ runtime->private_data = prtd;
+
+out:
+ return ret;
+}
+
+static int omap_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ kfree(runtime->private_data);
+ return 0;
+}
+
+static int omap_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+struct snd_pcm_ops omap_pcm_ops = {
+ .open = omap_pcm_open,
+ .close = omap_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = omap_pcm_hw_params,
+ .hw_free = omap_pcm_hw_free,
+ .prepare = omap_pcm_prepare,
+ .trigger = omap_pcm_trigger,
+ .pointer = omap_pcm_pointer,
+ .mmap = omap_pcm_mmap,
+};
+
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = omap_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &omap_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (dai->playback.channels_min) {
+ ret = omap_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = omap_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+
+out:
+ return ret;
+}
+
+struct snd_soc_platform omap_soc_platform = {
+ .name = "omap-pcm-audio",
+ .pcm_ops = &omap_pcm_ops,
+ .pcm_new = omap_pcm_new,
+ .pcm_free = omap_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(omap_soc_platform);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("OMAP PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
new file mode 100644
index 000000000000..e4369bdfd77d
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.h
@@ -0,0 +1,35 @@
+/*
+ * omap-pcm.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_PCM_H__
+#define __OMAP_PCM_H__
+
+struct omap_pcm_dma_data {
+ char *name; /* stream identifier */
+ int dma_req; /* DMA request line */
+ unsigned long port_addr; /* transmit/receive register */
+};
+
+extern struct snd_soc_platform omap_soc_platform;
+
+#endif
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-22 05:41:41 +0100