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-rw-r--r--sound/soc/omap/Kconfig23
-rw-r--r--sound/soc/omap/Makefile6
-rw-r--r--sound/soc/omap/ams-delta.c646
-rw-r--r--sound/soc/omap/n810.c7
-rw-r--r--sound/soc/omap/omap-mcbsp.c53
-rw-r--r--sound/soc/omap/omap-pcm.c59
-rw-r--r--sound/soc/omap/omap2evm.c2
-rw-r--r--sound/soc/omap/omap3beagle.c28
-rw-r--r--sound/soc/omap/omap3evm.c147
-rw-r--r--sound/soc/omap/omap3pandora.c4
-rw-r--r--sound/soc/omap/overo.c2
-rw-r--r--sound/soc/omap/sdp3430.c103
-rw-r--r--sound/soc/omap/zoom2.c314
13 files changed, 1333 insertions, 61 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 675732e724d5..2dee9839be86 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810
help
Say Y if you want to add support for SoC audio on Nokia N810.
+config SND_OMAP_SOC_AMS_DELTA
+ tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
+ depends on SND_OMAP_SOC && MACH_AMS_DELTA
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_CX20442
+ help
+ Say Y if you want to add support for SoC audio on Amstrad Delta.
+
config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
@@ -39,6 +47,14 @@ config SND_OMAP_SOC_OMAP2EVM
help
Say Y if you want to add support for SoC audio on the omap2evm board.
+config SND_OMAP_SOC_OMAP3EVM
+ tristate "SoC Audio support for OMAP3EVM board"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the omap3evm board.
+
config SND_OMAP_SOC_SDP3430
tristate "SoC Audio support for Texas Instruments SDP3430"
depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
@@ -64,4 +80,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE
help
Say Y if you want to add support for SoC audio on the Beagleboard.
+config SND_OMAP_SOC_ZOOM2
+ tristate "SoC Audio support for Zoom2"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for Soc audio on Zoom2 board.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0c9e4ac37660..02d69471dcb5 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,17 +7,23 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
+snd-soc-ams-delta-objs := ams-delta.o
snd-soc-osk5912-objs := osk5912.o
snd-soc-overo-objs := overo.o
snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-omap3evm-objs := omap3evm.o
snd-soc-sdp3430-objs := sdp3430.o
snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
+snd-soc-zoom2-objs := zoom2.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
+obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
new file mode 100644
index 000000000000..5a5166ac7279
--- /dev/null
+++ b/sound/soc/omap/ams-delta.c
@@ -0,0 +1,646 @@
+/*
+ * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
+ *
+ * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * Initially based on sound/soc/omap/osk5912.x
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * 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/gpio.h>
+#include <linux/spinlock.h>
+#include <linux/tty.h>
+
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <mach/board-ams-delta.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/cx20442.h"
+
+
+/* Board specific DAPM widgets */
+ const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+ /* Handset */
+ SND_SOC_DAPM_MIC("Mouthpiece", NULL),
+ SND_SOC_DAPM_HP("Earpiece", NULL),
+ /* Handsfree/Speakerphone */
+ SND_SOC_DAPM_MIC("Microphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* How they are connected to codec pins */
+static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
+ {"TELIN", NULL, "Mouthpiece"},
+ {"Earpiece", NULL, "TELOUT"},
+
+ {"MIC", NULL, "Microphone"},
+ {"Speaker", NULL, "SPKOUT"},
+};
+
+/*
+ * Controls, functional after the modem line discipline is activated.
+ */
+
+/* Virtual switch: audio input/output constellations */
+static const char *ams_delta_audio_mode[] =
+ {"Mixed", "Handset", "Handsfree", "Speakerphone"};
+
+/* Selection <-> pin translation */
+#define AMS_DELTA_MOUTHPIECE 0
+#define AMS_DELTA_EARPIECE 1
+#define AMS_DELTA_MICROPHONE 2
+#define AMS_DELTA_SPEAKER 3
+#define AMS_DELTA_AGC 4
+
+#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \
+ (1 << AMS_DELTA_MICROPHONE))
+#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \
+ (1 << AMS_DELTA_EARPIECE))
+#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \
+ (1 << AMS_DELTA_SPEAKER))
+#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
+
+unsigned short ams_delta_audio_mode_pins[] = {
+ AMS_DELTA_MIXED,
+ AMS_DELTA_HANDSET,
+ AMS_DELTA_HANDSFREE,
+ AMS_DELTA_SPEAKERPHONE,
+};
+
+static unsigned short ams_delta_audio_agc;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
+ unsigned short pins;
+ int pin, changed = 0;
+
+ /* Refuse any mode changes if we are not able to control the codec. */
+ if (!codec->control_data)
+ return -EUNATCH;
+
+ if (ucontrol->value.enumerated.item[0] >= control->max)
+ return -EINVAL;
+
+ mutex_lock(&codec->mutex);
+
+ /* Translate selection to bitmap */
+ pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
+
+ /* Setup pins after corresponding bits if changed */
+ pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Mouthpiece");
+ else
+ snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Earpiece");
+ else
+ snd_soc_dapm_disable_pin(codec, "Earpiece");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Microphone");
+ else
+ snd_soc_dapm_disable_pin(codec, "Microphone");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+ if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ }
+ pin = !!(pins & (1 << AMS_DELTA_AGC));
+ if (pin != ams_delta_audio_agc) {
+ ams_delta_audio_agc = pin;
+ changed = 1;
+ if (pin)
+ snd_soc_dapm_enable_pin(codec, "AGCIN");
+ else
+ snd_soc_dapm_disable_pin(codec, "AGCIN");
+ }
+ if (changed)
+ snd_soc_dapm_sync(codec);
+
+ mutex_unlock(&codec->mutex);
+
+ return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned short pins, mode;
+
+ pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
+ AMS_DELTA_MOUTHPIECE) |
+ (snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
+ AMS_DELTA_EARPIECE));
+ if (pins)
+ pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+ AMS_DELTA_MICROPHONE);
+ else
+ pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+ AMS_DELTA_MICROPHONE) |
+ (snd_soc_dapm_get_pin_status(codec, "Speaker") <<
+ AMS_DELTA_SPEAKER) |
+ (ams_delta_audio_agc << AMS_DELTA_AGC));
+
+ for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
+ if (pins == ams_delta_audio_mode_pins[mode])
+ break;
+
+ if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
+ return -EINVAL;
+
+ ucontrol->value.enumerated.item[0] = mode;
+
+ return 0;
+}
+
+static const struct soc_enum ams_delta_audio_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
+ ams_delta_audio_mode),
+};
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+ SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+ ams_delta_get_audio_mode, ams_delta_set_audio_mode),
+};
+
+/* Hook switch */
+static struct snd_soc_jack ams_delta_hook_switch;
+static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
+ {
+ .gpio = 4,
+ .name = "hook_switch",
+ .report = SND_JACK_HEADSET,
+ .invert = 1,
+ .debounce_time = 150,
+ }
+};
+
+/* After we are able to control the codec over the modem,
+ * the hook switch can be used for dynamic DAPM reconfiguration. */
+static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
+ /* Handset */
+ {
+ .pin = "Mouthpiece",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Earpiece",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ /* Handsfree */
+ {
+ .pin = "Microphone",
+ .mask = SND_JACK_MICROPHONE,
+ .invert = 1,
+ },
+ {
+ .pin = "Speaker",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1,
+ },
+};
+
+
+/*
+ * Modem line discipline, required for making above controls functional.
+ * Activated from userspace with ldattach, possibly invoked from udev rule.
+ */
+
+/* To actually apply any modem controlled configuration changes to the codec,
+ * we must connect codec DAI pins to the modem for a moment. Be carefull not
+ * to interfere with our digital mute function that shares the same hardware. */
+static struct timer_list cx81801_timer;
+static bool cx81801_cmd_pending;
+static bool ams_delta_muted;
+static DEFINE_SPINLOCK(ams_delta_lock);
+
+static void cx81801_timeout(unsigned long data)
+{
+ int muted;
+
+ spin_lock(&ams_delta_lock);
+ cx81801_cmd_pending = 0;
+ muted = ams_delta_muted;
+ spin_unlock(&ams_delta_lock);
+
+ /* Reconnect the codec DAI back from the modem to the CPU DAI
+ * only if digital mute still off */
+ if (!muted)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
+}
+
+/* Line discipline .open() */
+static int cx81801_open(struct tty_struct *tty)
+{
+ return v253_ops.open(tty);
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+ struct snd_soc_codec *codec = tty->disc_data;
+
+ del_timer_sync(&cx81801_timer);
+
+ v253_ops.close(tty);
+
+ /* Prevent the hook switch from further changing the DAPM pins */
+ INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+ /* Revert back to default audio input/output constellation */
+ snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ snd_soc_dapm_enable_pin(codec, "Earpiece");
+ snd_soc_dapm_enable_pin(codec, "Microphone");
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(codec, "AGCIN");
+ snd_soc_dapm_sync(codec);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+ cx81801_close(tty);
+ return 0;
+}
+
+/* Line discipline .recieve_buf() */
+static void cx81801_receive(struct tty_struct *tty,
+ const unsigned char *cp, char *fp, int count)
+{
+ struct snd_soc_codec *codec = tty->disc_data;
+ const unsigned char *c;
+ int apply, ret;
+
+ if (!codec->control_data) {
+ /* First modem response, complete setup procedure */
+
+ /* Initialize timer used for config pulse generation */
+ setup_timer(&cx81801_timer, cx81801_timeout, 0);
+
+ v253_ops.receive_buf(tty, cp, fp, count);
+
+ /* Link hook switch to DAPM pins */
+ ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_pins),
+ ams_delta_hook_switch_pins);
+ if (ret)
+ dev_warn(codec->socdev->card->dev,
+ "Failed to link hook switch to DAPM pins, "
+ "will continue with hook switch unlinked.\n");
+
+ return;
+ }
+
+ v253_ops.receive_buf(tty, cp, fp, count);
+
+ for (c = &cp[count - 1]; c >= cp; c--) {
+ if (*c != '\r')
+ continue;
+ /* Complete modem response received, apply config to codec */
+
+ spin_lock_bh(&ams_delta_lock);
+ mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
+ apply = !ams_delta_muted && !cx81801_cmd_pending;
+ cx81801_cmd_pending = 1;
+ spin_unlock_bh(&ams_delta_lock);
+
+ /* Apply config pulse by connecting the codec to the modem
+ * if not already done */
+ if (apply)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+ AMS_DELTA_LATCH2_MODEM_CODEC);
+ break;
+ }
+}
+
+/* Line discipline .write_wakeup() */
+static void cx81801_wakeup(struct tty_struct *tty)
+{
+ v253_ops.write_wakeup(tty);
+}
+
+static struct tty_ldisc_ops cx81801_ops = {
+ .magic = TTY_LDISC_MAGIC,
+ .name = "cx81801",
+ .owner = THIS_MODULE,
+ .open = cx81801_open,
+ .close = cx81801_close,
+ .hangup = cx81801_hangup,
+ .receive_buf = cx81801_receive,
+ .write_wakeup = cx81801_wakeup,
+};
+
+
+/*
+ * Even if not very usefull, the sound card can still work without any of the
+ * above functonality activated. You can still control its audio input/output
+ * constellation and speakerphone gain from userspace by issueing AT commands
+ * over the modem port.
+ */
+
+static int ams_delta_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* Set cpu DAI configuration */
+ return snd_soc_dai_set_fmt(rtd->dai->cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+}
+
+static struct snd_soc_ops ams_delta_ops = {
+ .hw_params = ams_delta_hw_params,
+};
+
+
+/* Board specific codec bias level control */
+static int ams_delta_set_bias_level(struct snd_soc_card *card,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_codec *codec = card->codec;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+ AMS_DELTA_LATCH2_MODEM_NRESET);
+ break;
+ case SND_SOC_BIAS_OFF:
+ if (codec->bias_level != SND_SOC_BIAS_OFF)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+ 0);
+ }
+ codec->bias_level = level;
+
+ return 0;
+}
+
+/* Digital mute implemented using modem/CPU multiplexer.
+ * Shares hardware with codec config pulse generation */
+static bool ams_delta_muted = 1;
+
+static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ int apply;
+
+ if (ams_delta_muted == mute)
+ return 0;
+
+ spin_lock_bh(&ams_delta_lock);
+ ams_delta_muted = mute;
+ apply = !cx81801_cmd_pending;
+ spin_unlock_bh(&ams_delta_lock);
+
+ if (apply)
+ ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+ mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
+ return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static struct snd_soc_dai_ops ams_delta_dai_ops = {
+ .digital_mute = ams_delta_digital_mute,
+};
+
+/* Will be used if the codec ever has its own digital_mute function */
+static int ams_delta_startup(struct snd_pcm_substream *substream)
+{
+ return ams_delta_digital_mute(NULL, 0);
+}
+
+static void ams_delta_shutdown(struct snd_pcm_substream *substream)
+{
+ ams_delta_digital_mute(NULL, 1);
+}
+
+
+/*
+ * Card initialization
+ */
+
+static int ams_delta_cx20442_init(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dai *codec_dai = codec->dai;
+ struct snd_soc_card *card = codec->socdev->card;
+ int ret;
+ /* Codec is ready, now add/activate board specific controls */
+
+ /* Set up digital mute if not provided by the codec */
+ if (!codec_dai->ops) {
+ codec_dai->ops = &ams_delta_dai_ops;
+ } else if (!codec_dai->ops->digital_mute) {
+ codec_dai->ops->digital_mute = ams_delta_digital_mute;
+ } else {
+ ams_delta_ops.startup = ams_delta_startup;
+ ams_delta_ops.shutdown = ams_delta_shutdown;
+ }
+
+ /* Set codec bias level */
+ ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+
+ /* Add hook switch - can be used to control the codec from userspace
+ * even if line discipline fails */
+ ret = snd_soc_jack_new(card, "hook_switch",
+ SND_JACK_HEADSET, &ams_delta_hook_switch);
+ if (ret)
+ dev_warn(card->dev,
+ "Failed to allocate resources for hook switch, "
+ "will continue without one.\n");
+ else {
+ ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios);
+ if (ret)
+ dev_warn(card->dev,
+ "Failed to set up hook switch GPIO line, "
+ "will continue with hook switch inactive.\n");
+ }
+
+ /* Register optional line discipline for over the modem control */
+ ret = tty_register_ldisc(N_V253, &cx81801_ops);
+ if (ret) {
+ dev_warn(card->dev,
+ "Failed to register line discipline, "
+ "will continue without any controls.\n");
+ return 0;
+ }
+
+ /* Add board specific DAPM widgets and routes */
+ ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+ ARRAY_SIZE(ams_delta_dapm_widgets));
+ if (ret) {
+ dev_warn(card->dev,
+ "Failed to register DAPM controls, "
+ "will continue without any.\n");
+ return 0;
+ }
+
+ ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+ ARRAY_SIZE(ams_delta_audio_map));
+ if (ret) {
+ dev_warn(card->dev,
+ "Failed to set up DAPM routes, "
+ "will continue with codec default map.\n");
+ return 0;
+ }
+
+ /* Set up initial pin constellation */
+ snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+ snd_soc_dapm_enable_pin(codec, "Earpiece");
+ snd_soc_dapm_enable_pin(codec, "Microphone");
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+ snd_soc_dapm_disable_pin(codec, "AGCIN");
+ snd_soc_dapm_disable_pin(codec, "AGCOUT");
+ snd_soc_dapm_sync(codec);
+
+ /* Add virtual switch */
+ ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+ ARRAY_SIZE(ams_delta_audio_controls));
+ if (ret)
+ dev_warn(card->dev,
+ "Failed to register audio mode control, "
+ "will continue without it.\n");
+
+ return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+ .name = "CX20442",
+ .stream_name = "CX20442",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &cx20442_dai,
+ .init = ams_delta_cx20442_init,
+ .ops = &ams_delta_ops,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+ .name = "AMS_DELTA",
+ .platform = &omap_soc_platform,
+ .dai_link = &ams_delta_dai_link,
+ .num_links = 1,
+ .set_bias_level = ams_delta_set_bias_level,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device ams_delta_snd_soc_device = {
+ .card = &ams_delta_audio_card,
+ .codec_dev = &cx20442_codec_dev,
+};
+
+/* Module init/exit */
+static struct platform_device *ams_delta_audio_platform_device;
+static struct platform_device *cx20442_platform_device;
+
+static int __init ams_delta_module_init(void)
+{
+ int ret;
+
+ if (!(machine_is_ams_delta()))
+ return -ENODEV;
+
+ ams_delta_audio_platform_device =
+ platform_device_alloc("soc-audio", -1);
+ if (!ams_delta_audio_platform_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(ams_delta_audio_platform_device,
+ &ams_delta_snd_soc_device);
+ ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev;
+ *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1;
+
+ ret = platform_device_add(ams_delta_audio_platform_device);
+ if (ret)
+ goto err;
+
+ /*
+ * Codec platform device could be registered from elsewhere (board?),
+ * but I do it here as it makes sense only if used with the card.
+ */
+ cx20442_platform_device = platform_device_register_simple("cx20442",
+ -1, NULL, 0);
+ return 0;
+err:
+ platform_device_put(ams_delta_audio_platform_device);
+ return ret;
+}
+module_init(ams_delta_module_init);
+
+static void __exit ams_delta_module_exit(void)
+{
+ struct snd_soc_codec *codec;
+ struct tty_struct *tty;
+
+ if (ams_delta_audio_card.codec) {
+ codec = ams_delta_audio_card.codec;
+
+ if (codec->control_data) {
+ tty = codec->control_data;
+
+ tty_hangup(tty);
+ }
+ }
+
+ if (tty_unregister_ldisc(N_V253) != 0)
+ dev_warn(&ams_delta_audio_platform_device->dev,
+ "failed to unregister V253 line discipline\n");
+
+ snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios);
+
+ /* Keep modem power on */
+ ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+
+ platform_device_unregister(cx20442_platform_device);
+ platform_device_unregister(ams_delta_audio_platform_device);
+}
+module_exit(ams_delta_module_exit);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 91ef17992de5..b60b1dfbc435 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -383,10 +383,9 @@ static int __init n810_soc_init(void)
clk_set_parent(sys_clkout2_src, func96m_clk);
clk_set_rate(sys_clkout2, 12000000);
- if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
- BUG();
- if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
- BUG();
+ BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+ (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+
gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 912614283848..6a837ffd5d0b 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -183,21 +183,21 @@ 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_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
- int err = 0;
+ int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
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);
+ mcbsp_data->active++;
+ omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
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);
+ omap_mcbsp_stop(mcbsp_data->bus_id, play, !play);
+ mcbsp_data->active--;
break;
default:
err = -EINVAL;
@@ -215,8 +215,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
- int wlen, channels;
+ int wlen, channels, wpf;
unsigned long port;
+ unsigned int format;
if (cpu_class_is_omap1()) {
dma = omap1_dma_reqs[bus_id][substream->stream];
@@ -244,18 +245,24 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
return 0;
}
- channels = params_channels(params);
+ format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+ wpf = channels = params_channels(params);
switch (channels) {
case 2:
- /* Use dual-phase frames */
- regs->rcr2 |= RPHASE;
- regs->xcr2 |= XPHASE;
+ if (format == SND_SOC_DAIFMT_I2S) {
+ /* Use dual-phase frames */
+ regs->rcr2 |= RPHASE;
+ regs->xcr2 |= XPHASE;
+ /* Set 1 word per (McBSP) frame for phase1 and phase2 */
+ wpf--;
+ regs->rcr2 |= RFRLEN2(wpf - 1);
+ regs->xcr2 |= XFRLEN2(wpf - 1);
+ }
case 1:
- /* Set 1 word per (McBSP) frame */
- regs->rcr2 |= RFRLEN2(1 - 1);
- regs->rcr1 |= RFRLEN1(1 - 1);
- regs->xcr2 |= XFRLEN2(1 - 1);
- regs->xcr1 |= XFRLEN1(1 - 1);
+ case 4:
+ /* Set word per (McBSP) frame for phase1 */
+ regs->rcr1 |= RFRLEN1(wpf - 1);
+ regs->xcr1 |= XFRLEN1(wpf - 1);
break;
default:
/* Unsupported number of channels */
@@ -277,11 +284,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
}
/* Set FS period and length in terms of bit clock periods */
- switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ switch (format) {
case SND_SOC_DAIFMT_I2S:
- regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr2 |= FPER(wlen * channels - 1);
regs->srgr1 |= FWID(wlen - 1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
regs->srgr2 |= FPER(wlen * channels - 1);
regs->srgr1 |= FWID(0);
@@ -326,6 +334,13 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* 1-bit data delay */
+ regs->rcr2 |= RDATDLY(1);
+ regs->xcr2 |= XDATDLY(1);
+ /* Invert FS polarity configuration */
+ temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+ break;
case SND_SOC_DAIFMT_DSP_B:
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
@@ -492,13 +507,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
.id = (link_id), \
.playback = { \
.channels_min = 1, \
- .channels_max = 2, \
+ .channels_max = 4, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
.capture = { \
.channels_min = 1, \
- .channels_max = 2, \
+ .channels_max = 4, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 07cf7f46b584..12e14c01068e 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -59,16 +59,31 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
struct omap_runtime_data *prtd = runtime->private_data;
unsigned long flags;
- if (cpu_is_omap1510()) {
+ if ((cpu_is_omap1510()) &&
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) {
/*
- * OMAP1510 doesn't support DMA chaining so have to restart
- * the transfer after all periods are transferred
+ * OMAP1510 doesn't fully support DMA progress counter
+ * and there is no software emulation implemented yet,
+ * so have to maintain our own playback progress counter
+ * that can be used by omap_pcm_pointer() instead.
*/
spin_lock_irqsave(&prtd->lock, flags);
+ if ((stat == OMAP_DMA_LAST_IRQ) &&
+ (prtd->period_index == runtime->periods - 1)) {
+ /* we are in sync, do nothing */
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return;
+ }
if (prtd->period_index >= 0) {
- if (++prtd->period_index == runtime->periods) {
+ if (stat & OMAP_DMA_BLOCK_IRQ) {
+ /* end of buffer reached, loop back */
+ prtd->period_index = 0;
+ } else if (stat & OMAP_DMA_LAST_IRQ) {
+ /* update the counter for the last period */
+ prtd->period_index = runtime->periods - 1;
+ } else if (++prtd->period_index >= runtime->periods) {
+ /* end of buffer missed? loop back */
prtd->period_index = 0;
- omap_start_dma(prtd->dma_ch);
}
}
spin_unlock_irqrestore(&prtd->lock, flags);
@@ -87,8 +102,10 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
int err = 0;
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
if (!dma_data)
- return -ENODEV;
+ return 0;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
@@ -98,7 +115,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
prtd->dma_data = dma_data;
err = omap_request_dma(dma_data->dma_req, dma_data->name,
omap_pcm_dma_irq, substream, &prtd->dma_ch);
- if (!err && !cpu_is_omap1510()) {
+ if (!err) {
/*
* Link channel with itself so DMA doesn't need any
* reprogramming while looping the buffer
@@ -117,8 +134,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
if (prtd->dma_data == NULL)
return 0;
- if (!cpu_is_omap1510())
- omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
+ omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
omap_free_dma(prtd->dma_ch);
prtd->dma_data = NULL;
@@ -134,6 +150,11 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
struct omap_pcm_dma_data *dma_data = prtd->dma_data;
struct omap_dma_channel_params dma_params;
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!prtd->dma_data)
+ return 0;
+
memset(&dma_params, 0, sizeof(dma_params));
/*
* Note: Regardless of interface data formats supported by OMAP McBSP
@@ -167,7 +188,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
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);
+ if ((cpu_is_omap1510()) &&
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+ omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
+ OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
+ else
+ omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
return 0;
}
@@ -209,12 +235,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
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
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+ offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+ } else if (!(cpu_is_omap1510())) {
+ ptr = omap_get_dma_src_pos(prtd->dma_ch);
+ offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+ } else
+ offset = prtd->period_index * runtime->period_size;
- offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
if (offset >= runtime->buffer_size)
offset = 0;
@@ -320,7 +349,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
}
}
-int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
struct snd_pcm *pcm)
{
int ret = 0;
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 0c2322dcf02a..027e1a40f8a1 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = {
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap2evm_ops,
};
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index fd24a4acd2f5..b0cff9f33b7e 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -41,23 +41,33 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int fmt;
int ret;
+ switch (params_channels(params)) {
+ case 2: /* Stereo I2S mode */
+ fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+ break;
+ case 4: /* Four channel TDM mode */
+ fmt = SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+ break;
+ default:
+ return -EINVAL;
+ }
+
/* Set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
if (ret < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return ret;
}
/* Set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
if (ret < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
return ret;
@@ -83,7 +93,7 @@ static struct snd_soc_dai_link omap3beagle_dai = {
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3beagle_ops,
};
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
new file mode 100644
index 000000000000..9114c263077b
--- /dev/null
+++ b/sound/soc/omap/omap3evm.c
@@ -0,0 +1,147 @@
+/*
+ * omap3evm.c -- ALSA SoC support for OMAP3 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2008 Texas Instruments, Incorporated
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3evm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "Can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "Can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "Can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap3evm_ops = {
+ .hw_params = omap3evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3evm_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+ .ops = &omap3evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3evm = {
+ .name = "omap3evm",
+ .platform = &omap_soc_platform,
+ .dai_link = &omap3evm_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3evm_snd_devdata = {
+ .card = &snd_soc_omap3evm,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3evm_snd_device;
+
+static int __init omap3evm_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap3evm()) {
+ pr_err("Not OMAP3 EVM!\n");
+ return -ENODEV;
+ }
+ pr_info("OMAP3 EVM SoC init\n");
+
+ omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!omap3evm_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata);
+ omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev;
+ *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1;
+
+ ret = platform_device_add(omap3evm_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(omap3evm_snd_device);
+
+ return ret;
+}
+
+static void __exit omap3evm_soc_exit(void)
+{
+ platform_device_unregister(omap3evm_snd_device);
+}
+
+module_init(omap3evm_soc_init);
+module_exit(omap3evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index fe282d4ef422..ad219aaf7cb8 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
.name = "PCM1773",
.stream_name = "HiFi Out",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_out_ops,
.init = omap3pandora_out_init,
}, {
.name = "TWL4030",
.stream_name = "Line/Mic In",
.cpu_dai = &omap_mcbsp_dai[1],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_in_ops,
.init = omap3pandora_in_init,
}
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index a72dc4e159e5..ec4f8fd8b3a2 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = {
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &overo_ops,
};
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 10f1c867f11d..f7e5b7488c35 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -24,6 +24,7 @@
#include <linux/clk.h>
#include <linux/platform_device.h>
+#include <linux/i2c/twl4030.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -39,6 +40,9 @@
#include "omap-pcm.h"
#include "../codecs/twl4030.h"
+#define TWL4030_INTBR_PMBR1 0x0D
+#define EXTMUTE(value) (value << 2)
+
static struct snd_soc_card snd_soc_sdp3430;
static int sdp3430_hw_params(struct snd_pcm_substream *substream,
@@ -84,6 +88,49 @@ static struct snd_soc_ops sdp3430_ops = {
.hw_params = sdp3430_hw_params,
};
+static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops sdp3430_voice_ops = {
+ .hw_params = sdp3430_hw_voice_params,
+};
+
/* Headset jack */
static struct snd_soc_jack hs_jack;
@@ -192,28 +239,59 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec)
return ret;
}
+static int sdp3430_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+ unsigned short reg;
+
+ /* Enable voice interface */
+ reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+ reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+ codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+ return 0;
+}
+
+
/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp3430_dai = {
- .name = "TWL4030",
- .stream_name = "TWL4030",
- .cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
- .init = sdp3430_twl4030_init,
- .ops = &sdp3430_ops,
+static struct snd_soc_dai_link sdp3430_dai[] = {
+ {
+ .name = "TWL4030 I2S",
+ .stream_name = "TWL4030 Audio",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+ .init = sdp3430_twl4030_init,
+ .ops = &sdp3430_ops,
+ },
+ {
+ .name = "TWL4030 PCM",
+ .stream_name = "TWL4030 Voice",
+ .cpu_dai = &omap_mcbsp_dai[1],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+ .init = sdp3430_twl4030_voice_init,
+ .ops = &sdp3430_voice_ops,
+ },
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_sdp3430 = {
.name = "SDP3430",
.platform = &omap_soc_platform,
- .dai_link = &sdp3430_dai,
- .num_links = 1,
+ .dai_link = sdp3430_dai,
+ .num_links = ARRAY_SIZE(sdp3430_dai),
+};
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+ .ramp_delay_value = 3,
+ .sysclk = 26000,
+ .hs_extmute = 1,
};
/* Audio subsystem */
static struct snd_soc_device sdp3430_snd_devdata = {
.card = &snd_soc_sdp3430,
.codec_dev = &soc_codec_dev_twl4030,
+ .codec_data = &twl4030_setup,
};
static struct platform_device *sdp3430_snd_device;
@@ -236,7 +314,12 @@ static int __init sdp3430_soc_init(void)
platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
- *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+ *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
+
+ /* Set TWL4030 GPIO6 as EXTMUTE signal */
+ twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02),
+ TWL4030_MODULE_INTBR);
ret = platform_device_add(sdp3430_snd_device);
if (ret)
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
new file mode 100644
index 000000000000..f90b45f56220
--- /dev/null
+++ b/sound/soc/omap/zoom2.c
@@ -0,0 +1,314 @@
+/*
+ * zoom2.c -- SoC audio for Zoom2
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15)
+#define ZOOM2_HEADSET_EXTMUTE_GPIO 153
+
+static int zoom2_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops zoom2_ops = {
+ .hw_params = zoom2_hw_params,
+};
+
+static int zoom2_hw_voice_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops zoom2_voice_ops = {
+ .hw_params = zoom2_hw_voice_params,
+};
+
+/* Zoom2 machine DAPM */
+static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Ext Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* External Mics: MAINMIC, SUBMIC with bias*/
+ {"MAINMIC", NULL, "Mic Bias 1"},
+ {"SUBMIC", NULL, "Mic Bias 2"},
+ {"Mic Bias 1", NULL, "Ext Mic"},
+ {"Mic Bias 2", NULL, "Ext Mic"},
+
+ /* External Speakers: HFL, HFR */
+ {"Ext Spk", NULL, "HFL"},
+ {"Ext Spk", NULL, "HFR"},
+
+ /* Headset Stereophone: HSOL, HSOR */
+ {"Headset Stereophone", NULL, "HSOL"},
+ {"Headset Stereophone", NULL, "HSOR"},
+
+ /* Headset Mic: HSMIC with bias */
+ {"HSMIC", NULL, "Headset Mic Bias"},
+ {"Headset Mic Bias", NULL, "Headset Mic"},
+
+ /* Aux In: AUXL, AUXR */
+ {"Aux In", NULL, "AUXL"},
+ {"Aux In", NULL, "AUXR"},
+};
+
+static int zoom2_twl4030_init(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ /* Add Zoom2 specific widgets */
+ ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
+ ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
+ if (ret)
+ return ret;
+
+ /* Set up Zoom2 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* Zoom2 connected pins */
+ snd_soc_dapm_enable_pin(codec, "Ext Mic");
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+ snd_soc_dapm_enable_pin(codec, "Aux In");
+
+ /* TWL4030 not connected pins */
+ snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+ snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+ snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+ snd_soc_dapm_nc_pin(codec, "OUTL");
+ snd_soc_dapm_nc_pin(codec, "OUTR");
+ snd_soc_dapm_nc_pin(codec, "EARPIECE");
+ snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+ snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+ snd_soc_dapm_nc_pin(codec, "CARKITL");
+ snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+ ret = snd_soc_dapm_sync(codec);
+
+ return ret;
+}
+
+static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+ unsigned short reg;
+
+ /* Enable voice interface */
+ reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+ reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+ codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link zoom2_dai[] = {
+ {
+ .name = "TWL4030 I2S",
+ .stream_name = "TWL4030 Audio",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+ .init = zoom2_twl4030_init,
+ .ops = &zoom2_ops,
+ },
+ {
+ .name = "TWL4030 PCM",
+ .stream_name = "TWL4030 Voice",
+ .cpu_dai = &omap_mcbsp_dai[1],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+ .init = zoom2_twl4030_voice_init,
+ .ops = &zoom2_voice_ops,
+ },
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_zoom2 = {
+ .name = "Zoom2",
+ .platform = &omap_soc_platform,
+ .dai_link = zoom2_dai,
+ .num_links = ARRAY_SIZE(zoom2_dai),
+};
+
+/* EXTMUTE callback function */
+void zoom2_set_hs_extmute(int mute)
+{
+ gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute);
+}
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+ .ramp_delay_value = 3, /* 161 ms */
+ .sysclk = 26000,
+ .hs_extmute = 1,
+ .set_hs_extmute = zoom2_set_hs_extmute,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device zoom2_snd_devdata = {
+ .card = &snd_soc_zoom2,
+ .codec_dev = &soc_codec_dev_twl4030,
+ .codec_data = &twl4030_setup,
+};
+
+static struct platform_device *zoom2_snd_device;
+
+static int __init zoom2_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap_zoom2()) {
+ pr_debug("Not Zoom2!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO "Zoom2 SoC init\n");
+
+ zoom2_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!zoom2_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata);
+ zoom2_snd_devdata.dev = &zoom2_snd_device->dev;
+ *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+ *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
+
+ ret = platform_device_add(zoom2_snd_device);
+ if (ret)
+ goto err1;
+
+ BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
+ gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
+
+ BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
+ gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(zoom2_snd_device);
+
+ return ret;
+}
+module_init(zoom2_soc_init);
+
+static void __exit zoom2_soc_exit(void)
+{
+ gpio_free(ZOOM2_HEADSET_MUX_GPIO);
+ gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
+
+ platform_device_unregister(zoom2_snd_device);
+}
+module_exit(zoom2_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC Zoom2");
+MODULE_LICENSE("GPL");
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-21 05:17:30 +0000