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-rw-r--r--sound/soc/omap/Kconfig15
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/ams-delta.c646
-rw-r--r--sound/soc/omap/n810.c12
-rw-r--r--sound/soc/omap/omap-mcbsp.c123
-rw-r--r--sound/soc/omap/omap-mcbsp.h4
-rw-r--r--sound/soc/omap/omap-pcm.c53
-rw-r--r--sound/soc/omap/omap-pcm.h2
-rw-r--r--sound/soc/omap/sdp3430.c18
-rw-r--r--sound/soc/omap/zoom2.c314
10 files changed, 1159 insertions, 32 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index b771238662b6..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
@@ -72,4 +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 a37f49862389..02d69471dcb5 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
# OMAP Machine Support
snd-soc-n810-objs := n810.o
+snd-soc-ams-delta-objs := ams-delta.o
snd-soc-osk5912-objs := osk5912.o
snd-soc-overo-objs := overo.o
snd-soc-omap2evm-objs := omap2evm.o
@@ -14,8 +15,10 @@ snd-soc-omap3evm-objs := omap3evm.o
snd-soc-sdp3430-objs := sdp3430.o
snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
+snd-soc-zoom2-objs := zoom2.o
obj-$(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
@@ -23,3 +26,4 @@ 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 b60b1dfbc435..0a505938e42b 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -22,6 +22,7 @@
*/
#include <linux/clk.h>
+#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -322,8 +323,6 @@ static struct snd_soc_card snd_soc_n810 = {
/* Audio private data */
static struct aic3x_setup_data n810_aic33_setup = {
- .i2c_bus = 2,
- .i2c_address = 0x18,
.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
};
@@ -337,6 +336,13 @@ static struct snd_soc_device n810_snd_devdata = {
static struct platform_device *n810_snd_device;
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+ { I2C_BOARD_INFO("tlv320aic3x", 0x1b), }
+};
+
static int __init n810_soc_init(void)
{
int err;
@@ -345,6 +351,8 @@ static int __init n810_soc_init(void)
if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
return -ENODEV;
+ i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device));
+
n810_snd_device = platform_device_alloc("soc-audio", -1);
if (!n810_snd_device)
return -ENOMEM;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index a5d46a7b196a..3341f49402ca 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -139,27 +139,67 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
static const unsigned long omap34xx_mcbsp_port[][2] = {};
#endif
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+{
+ 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 dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
+ int samples;
+
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ samples = snd_pcm_lib_period_bytes(substream) >> 1;
+ else
+ samples = 1;
+
+ /* Configure McBSP internal buffer usage */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
+ else
+ omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
+}
+
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
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 bus_id = mcbsp_data->bus_id;
int err = 0;
- if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+ if (!cpu_dai->active)
+ err = omap_mcbsp_request(bus_id);
+
+ if (cpu_is_omap343x()) {
+ int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
+ int max_period;
+
/*
* McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
* Set constraint for minimum buffer size to the same than FIFO
* size in order to avoid underruns in playback startup because
* HW is keeping the DMA request active until FIFO is filled.
*/
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
- }
+ if (bus_id == 1)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ 4096, UINT_MAX);
- if (!cpu_dai->active)
- err = omap_mcbsp_request(mcbsp_data->bus_id);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
+ else
+ max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
+
+ max_period++;
+ max_period <<= 1;
+
+ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ 32, max_period);
+ }
return err;
}
@@ -183,21 +223,21 @@ 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;
@@ -215,7 +255,7 @@ 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, wpf;
+ int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
unsigned long port;
unsigned int format;
@@ -231,6 +271,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
+ omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
+ omap_mcbsp_set_threshold;
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+ MCBSP_DMA_MODE_THRESHOLD)
+ sync_mode = OMAP_DMA_SYNC_FRAME;
} else {
return -ENODEV;
}
@@ -238,6 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
substream->stream ? "Audio Capture" : "Audio Playback";
omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
+ omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
if (mcbsp_data->configured) {
@@ -321,11 +368,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* Generic McBSP register settings */
regs->spcr2 |= XINTM(3) | FREE;
regs->spcr1 |= RINTM(3);
- regs->rcr2 |= RFIG;
- regs->xcr2 |= XFIG;
+ /* RFIG and XFIG are not defined in 34xx */
+ if (!cpu_is_omap34xx()) {
+ regs->rcr2 |= RFIG;
+ regs->xcr2 |= XFIG;
+ }
if (cpu_is_omap2430() || cpu_is_omap34xx()) {
- regs->xccr = DXENDLY(1) | XDMAEN;
- regs->rccr = RFULL_CYCLE | RDMAEN;
+ regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
+ regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -462,6 +512,40 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
return 0;
}
+static int omap_mcbsp_dai_set_rcvr_src(struct omap_mcbsp_data *mcbsp_data,
+ int clk_id)
+{
+ int sel_bit, set = 0;
+ u16 reg = OMAP2_CONTROL_DEVCONF0;
+
+ if (cpu_class_is_omap1())
+ return -EINVAL; /* TODO: Can this be implemented for OMAP1? */
+ if (mcbsp_data->bus_id != 0)
+ return -EINVAL;
+
+ switch (clk_id) {
+ case OMAP_MCBSP_CLKR_SRC_CLKX:
+ set = 1;
+ case OMAP_MCBSP_CLKR_SRC_CLKR:
+ sel_bit = 3;
+ break;
+ case OMAP_MCBSP_FSR_SRC_FSX:
+ set = 1;
+ case OMAP_MCBSP_FSR_SRC_FSR:
+ sel_bit = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (set)
+ 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_dai *cpu_dai,
int clk_id, unsigned int freq,
int dir)
@@ -484,6 +568,13 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
case OMAP_MCBSP_SYSCLK_CLKR_EXT:
regs->pcr0 |= SCLKME;
break;
+
+ case OMAP_MCBSP_CLKR_SRC_CLKR:
+ case OMAP_MCBSP_CLKR_SRC_CLKX:
+ case OMAP_MCBSP_FSR_SRC_FSR:
+ case OMAP_MCBSP_FSR_SRC_FSX:
+ err = omap_mcbsp_dai_set_rcvr_src(mcbsp_data, clk_id);
+ break;
default:
err = -ENODEV;
}
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index c8147aace813..647d2f981ab0 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -32,6 +32,10 @@ enum omap_mcbsp_clksrg_clk {
OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
+ OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */
+ OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */
+ OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */
+ OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */
};
/* McBSP dividers */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 84a1950880eb..5735945788bf 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);
@@ -100,7 +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
@@ -119,8 +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;
@@ -148,7 +162,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
*/
dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
dma_params.trigger = dma_data->dma_req;
- dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT;
+ dma_params.sync_mode = dma_data->sync_mode;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
@@ -174,7 +188,15 @@ 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);
+
+ omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
return 0;
}
@@ -183,6 +205,7 @@ 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;
+ struct omap_pcm_dma_data *dma_data = prtd->dma_data;
unsigned long flags;
int ret = 0;
@@ -192,6 +215,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->period_index = 0;
+ /* Configure McBSP internal buffer usage */
+ if (dma_data->set_threshold)
+ dma_data->set_threshold(substream);
+
omap_start_dma(prtd->dma_ch);
break;
@@ -288,7 +315,7 @@ static struct snd_pcm_ops omap_pcm_ops = {
.mmap = omap_pcm_mmap,
};
-static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
int stream)
@@ -330,7 +357,7 @@ 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;
@@ -338,7 +365,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
if (!card->dev->dma_mask)
card->dev->dma_mask = &omap_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
if (dai->playback.channels_min) {
ret = omap_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 8d9d26916b05..38a821dd4118 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,6 +29,8 @@ struct omap_pcm_dma_data {
char *name; /* stream identifier */
int dma_req; /* DMA request line */
unsigned long port_addr; /* transmit/receive register */
+ int sync_mode; /* DMA sync mode */
+ void (*set_threshold)(struct snd_pcm_substream *substream);
};
extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index b719e5db4f57..4a3f62d1f295 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 +40,11 @@
#include "omap-pcm.h"
#include "../codecs/twl4030.h"
+/* TWL4030 PMBR1 Register */
+#define TWL4030_INTBR_PMBR1 0x0D
+/* TWL4030 PMBR1 Register GPIO6 mux bit */
+#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2)
+
static struct snd_soc_card snd_soc_sdp3430;
static int sdp3430_hw_params(struct snd_pcm_substream *substream,
@@ -96,7 +102,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_DSP_A |
SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBS_CFM);
+ SND_SOC_DAIFMT_CBM_CFM);
if (ret) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return ret;
@@ -280,6 +286,7 @@ static struct snd_soc_card snd_soc_sdp3430 = {
static struct twl4030_setup_data twl4030_setup = {
.ramp_delay_value = 3,
.sysclk = 26000,
+ .hs_extmute = 1,
};
/* Audio subsystem */
@@ -294,6 +301,7 @@ static struct platform_device *sdp3430_snd_device;
static int __init sdp3430_soc_init(void)
{
int ret;
+ u8 pin_mux;
if (!machine_is_omap_3430sdp()) {
pr_debug("Not SDP3430!\n");
@@ -312,6 +320,14 @@ static int __init sdp3430_soc_init(void)
*(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_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+ TWL4030_INTBR_PMBR1);
+ pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
+ pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
+ twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+ TWL4030_INTBR_PMBR1);
+
ret = platform_device_add(sdp3430_snd_device);
if (ret)
goto err1;
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 2024-11-24 12:00:05 +0000