Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
authorLinus Torvalds <torvalds@linux-foundation.org>
Tue, 29 Apr 2008 16:38:52 +0000 (09:38 -0700)
committerLinus Torvalds <torvalds@linux-foundation.org>
Tue, 29 Apr 2008 16:38:52 +0000 (09:38 -0700)
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  [ALSA] soc - wm9712 - checkpatch fixes
  [ALSA] pcsp - Fix more dependency
  [ALSA] hda - Add support of Medion RIM 2150
  [ALSA] ASoC: Add drivers for the Texas Instruments OMAP processors
  [ALSA] ice1724 - Enable watermarks
  [ALSA] Add MPU401_INFO_NO_ACK bitflag

16 files changed:
Documentation/sound/alsa/ALSA-Configuration.txt
include/sound/mpu401.h
sound/drivers/Kconfig
sound/drivers/mpu401/mpu401_uart.c
sound/pci/hda/patch_realtek.c
sound/pci/ice1712/ice1724.c
sound/soc/Kconfig
sound/soc/Makefile
sound/soc/codecs/wm9712.c
sound/soc/omap/Kconfig [new file with mode: 0644]
sound/soc/omap/Makefile [new file with mode: 0644]
sound/soc/omap/n810.c [new file with mode: 0644]
sound/soc/omap/omap-mcbsp.c [new file with mode: 0644]
sound/soc/omap/omap-mcbsp.h [new file with mode: 0644]
sound/soc/omap/omap-pcm.c [new file with mode: 0644]
sound/soc/omap/omap-pcm.h [new file with mode: 0644]

index fd4c32a..0bbee38 100644 (file)
@@ -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
index 68b634b..1f1d53f 100644 (file)
@@ -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
index fe85af1..a78a8d0 100644 (file)
@@ -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
index 18cca24..2af0999 100644 (file)
@@ -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) {
index cdda64b..d9783a4 100644 (file)
@@ -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 },
index 4490422..6735090 100644 (file)
@@ -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
                }
        }
 
index a3b51df..18f28ac 100644 (file)
@@ -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"
index e489dbd..782db21 100644 (file)
@@ -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/
index d2d79e1..76c1e2d 100644 (file)
@@ -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 (file)
index 0000000..0230d83
--- /dev/null
@@ -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 (file)
index 0000000..d8d8d58
--- /dev/null
@@ -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 (file)
index 0000000..83b1eb4
--- /dev/null
@@ -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 (file)
index 0000000..40d87e6
--- /dev/null
@@ -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 (file)
index 0000000..9965fd4
--- /dev/null
@@ -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 (file)
index 0000000..6237020
--- /dev/null
@@ -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 (file)
index 0000000..e4369bd
--- /dev/null
@@ -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