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-rw-r--r--sound/soc/omap/Kconfig36
-rw-r--r--sound/soc/omap/Makefile5
-rw-r--r--sound/soc/omap/am3517evm.c3
-rw-r--r--sound/soc/omap/ams-delta.c136
-rw-r--r--sound/soc/omap/mcbsp.c55
-rw-r--r--sound/soc/omap/mcbsp.h7
-rw-r--r--sound/soc/omap/n810.c44
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c146
-rw-r--r--sound/soc/omap/omap-dmic.c60
-rw-r--r--sound/soc/omap/omap-hdmi-card.c4
-rw-r--r--sound/soc/omap/omap-hdmi.c41
-rw-r--r--sound/soc/omap/omap-mcbsp.c55
-rw-r--r--sound/soc/omap/omap-mcbsp.h2
-rw-r--r--sound/soc/omap/omap-mcpdm.c151
-rw-r--r--sound/soc/omap/omap-pcm.c141
-rw-r--r--sound/soc/omap/omap-pcm.h40
-rw-r--r--sound/soc/omap/omap-twl4030.c238
-rw-r--r--sound/soc/omap/omap3pandora.c52
-rw-r--r--sound/soc/omap/osk5912.c3
-rw-r--r--sound/soc/omap/rx51.c308
-rw-r--r--sound/soc/omap/sdp3430.c278
-rw-r--r--sound/soc/omap/zoom2.c207
22 files changed, 807 insertions, 1205 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 7048137f9a3..d44463a7b0f 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,7 +1,7 @@
config SND_OMAP_SOC
tristate "SoC Audio for the Texas Instruments OMAP chips"
- depends on ARCH_OMAP && DMA_OMAP
- select SND_SOC_DMAENGINE_PCM
+ depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST)
+ select SND_DMAENGINE_PCM
config SND_OMAP_SOC_DMIC
tristate
@@ -26,17 +26,18 @@ config SND_OMAP_SOC_N810
config SND_OMAP_SOC_RX51
tristate "SoC Audio support for Nokia RX-51"
- depends on SND_OMAP_SOC && MACH_NOKIA_RX51
+ depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C
select SND_OMAP_SOC_MCBSP
select SND_SOC_TLV320AIC3X
select SND_SOC_TPA6130A2
+ depends on GPIOLIB
help
Say Y if you want to add support for SoC audio on Nokia RX-51
hardware. This is also known as Nokia N900 product.
config SND_OMAP_SOC_AMS_DELTA
tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
- depends on SND_OMAP_SOC && MACH_AMS_DELTA
+ depends on SND_OMAP_SOC && MACH_AMS_DELTA && TTY
select SND_OMAP_SOC_MCBSP
select SND_SOC_CX20442
help
@@ -57,7 +58,7 @@ config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y if you want to add support for SoC audio on osk5912.
@@ -65,20 +66,11 @@ config SND_OMAP_SOC_AM3517EVM
tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
EVM.
-config SND_OMAP_SOC_SDP3430
- tristate "SoC Audio support for Texas Instruments SDP3430"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
- select SND_OMAP_SOC_MCBSP
- select SND_SOC_TWL4030
- help
- Say Y if you want to add support for SoC audio on Texas Instruments
- SDP3430.
-
config SND_OMAP_SOC_OMAP_TWL4030
tristate "SoC Audio support for TI SoC based boards with twl4030 codec"
depends on TWL4030_CORE && SND_OMAP_SOC
@@ -91,10 +83,12 @@ config SND_OMAP_SOC_OMAP_TWL4030
- Gumstix Overo or CompuLab CM-T35/CM-T3730
- IGEP v2
- OMAP3EVM
+ - SDP3430
+ - Zoom2
config SND_OMAP_SOC_OMAP_ABE_TWL6040
tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
- depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4
+ depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST)
select SND_OMAP_SOC_DMIC
select SND_OMAP_SOC_MCPDM
select SND_SOC_TWL6040
@@ -110,7 +104,7 @@ config SND_OMAP_SOC_OMAP_HDMI
tristate "SoC Audio support for Texas Instruments OMAP HDMI"
depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS
select SND_OMAP_SOC_HDMI
- select SND_SOC_OMAP_HDMI_CODEC
+ select SND_SOC_HDMI_CODEC
select OMAP4_DSS_HDMI_AUDIO
help
Say Y if you want to add support for SoC HDMI audio on Texas Instruments
@@ -123,11 +117,3 @@ config SND_OMAP_SOC_OMAP3_PANDORA
select SND_SOC_TWL4030
help
Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
-
-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 19637e55ea4..a725905b2c6 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -17,22 +17,17 @@ snd-soc-rx51-objs := rx51.o
snd-soc-ams-delta-objs := ams-delta.o
snd-soc-osk5912-objs := osk5912.o
snd-soc-am3517evm-objs := am3517evm.o
-snd-soc-sdp3430-objs := sdp3430.o
snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
snd-soc-omap-twl4030-objs := omap-twl4030.o
snd-soc-omap3pandora-objs := omap3pandora.o
-snd-soc-zoom2-objs := zoom2.o
snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
-obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
-obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index c1900b2a6f2..25a33e9d417 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -28,7 +28,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#include "../codecs/tlv320aic23.h"
@@ -78,7 +77,7 @@ static struct snd_soc_dai_link am3517evm_dai = {
.stream_name = "AIC23",
.cpu_dai_name = "omap-mcbsp.1",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.1",
.codec_name = "tlv320aic23-codec.2-001a",
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2600447fa74..0cc41f94de4 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -36,10 +36,8 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#include "../codecs/cx20442.h"
-
/* Board specific DAPM widgets */
static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
/* Handset */
@@ -91,73 +89,81 @@ static const unsigned short ams_delta_audio_mode_pins[] = {
static unsigned short ams_delta_audio_agc;
+/*
+ * Used for passing a codec structure pointer
+ * from the board initialization code to the tty line discipline.
+ */
+static struct snd_soc_codec *cx20442_codec;
+
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 snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &card->dapm;
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->hw_write)
+ if (!cx20442_codec->hw_write)
return -EUNATCH;
- if (ucontrol->value.enumerated.item[0] >= control->max)
+ if (ucontrol->value.enumerated.item[0] >= control->items)
return -EINVAL;
- mutex_lock(&codec->mutex);
+ snd_soc_dapm_mutex_lock(dapm);
/* 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(dapm, "Mouthpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece");
else
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
}
pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
else
- snd_soc_dapm_disable_pin(dapm, "Earpiece");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece");
}
pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Microphone");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
else
- snd_soc_dapm_disable_pin(dapm, "Microphone");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone");
}
pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "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(dapm, "AGCIN");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "AGCIN");
else
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
}
+
if (changed)
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
- mutex_unlock(&codec->mutex);
+ snd_soc_dapm_mutex_unlock(dapm);
return changed;
}
@@ -165,8 +171,8 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
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);
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &card->dapm;
unsigned short pins, mode;
pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
@@ -195,13 +201,11 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
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 SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
+ ams_delta_audio_mode);
static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
- SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+ SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum,
ams_delta_get_audio_mode, ams_delta_set_audio_mode),
};
@@ -271,12 +275,6 @@ static void cx81801_timeout(unsigned long data)
ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
}
-/*
- * Used for passing a codec structure pointer
- * from the board initialization code to the tty line discipline.
- */
-static struct snd_soc_codec *cx20442_codec;
-
/* Line discipline .open() */
static int cx81801_open(struct tty_struct *tty)
{
@@ -303,7 +301,7 @@ static int cx81801_open(struct tty_struct *tty)
static void cx81801_close(struct tty_struct *tty)
{
struct snd_soc_codec *codec = tty->disc_data;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dapm_context *dapm = &codec->card->dapm;
del_timer_sync(&cx81801_timer);
@@ -316,12 +314,17 @@ static void cx81801_close(struct tty_struct *tty)
v253_ops.close(tty);
/* Revert back to default audio input/output constellation */
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- snd_soc_dapm_enable_pin(dapm, "Microphone");
- snd_soc_dapm_disable_pin(dapm, "Speaker");
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
/* Line discipline .hangup() */
@@ -471,15 +474,14 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream)
static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dapm_context *dapm = &card->dapm;
int ret;
/* Codec is ready, now add/activate board specific controls */
/* Store a pointer to the codec structure for tty ldisc use */
- cx20442_codec = codec;
+ cx20442_codec = rtd->codec;
/* Set up digital mute if not provided by the codec */
if (!codec_dai->driver->ops) {
@@ -516,40 +518,20 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
- /* Add board specific DAPM widgets and routes */
- ret = snd_soc_dapm_new_controls(dapm, 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(dapm, 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(dapm, "Mouthpiece");
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- snd_soc_dapm_enable_pin(dapm, "Microphone");
snd_soc_dapm_disable_pin(dapm, "Speaker");
snd_soc_dapm_disable_pin(dapm, "AGCIN");
snd_soc_dapm_disable_pin(dapm, "AGCOUT");
- /* Add virtual switch */
- ret = snd_soc_add_codec_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;
+}
+
+static int ams_delta_card_remove(struct snd_soc_card *card)
+{
+ snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios);
return 0;
}
@@ -561,7 +543,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
.cpu_dai_name = "omap-mcbsp.1",
.codec_dai_name = "cx20442-voice",
.init = ams_delta_cx20442_init,
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.1",
.codec_name = "cx20442-codec",
.ops = &ams_delta_ops,
};
@@ -570,8 +552,16 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
static struct snd_soc_card ams_delta_audio_card = {
.name = "AMS_DELTA",
.owner = THIS_MODULE,
+ .remove = ams_delta_card_remove,
.dai_link = &ams_delta_dai_link,
.num_links = 1,
+
+ .controls = ams_delta_audio_controls,
+ .num_controls = ARRAY_SIZE(ams_delta_audio_controls),
+ .dapm_widgets = ams_delta_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ams_delta_dapm_widgets),
+ .dapm_routes = ams_delta_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(ams_delta_audio_map),
};
/* Module init/exit */
@@ -599,10 +589,6 @@ static int ams_delta_remove(struct platform_device *pdev)
dev_warn(&pdev->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);
-
snd_soc_unregister_card(card);
card->dev = NULL;
return 0;
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 285c8368cb4..86c75384c3c 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -36,10 +36,10 @@ static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
if (mcbsp->pdata->reg_size == 2) {
((u16 *)mcbsp->reg_cache)[reg] = (u16)val;
- __raw_writew((u16)val, addr);
+ writew_relaxed((u16)val, addr);
} else {
((u32 *)mcbsp->reg_cache)[reg] = val;
- __raw_writel(val, addr);
+ writel_relaxed(val, addr);
}
}
@@ -48,22 +48,22 @@ static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache)
void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step;
if (mcbsp->pdata->reg_size == 2) {
- return !from_cache ? __raw_readw(addr) :
+ return !from_cache ? readw_relaxed(addr) :
((u16 *)mcbsp->reg_cache)[reg];
} else {
- return !from_cache ? __raw_readl(addr) :
+ return !from_cache ? readl_relaxed(addr) :
((u32 *)mcbsp->reg_cache)[reg];
}
}
static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
{
- __raw_writel(val, mcbsp->st_data->io_base_st + reg);
+ writel_relaxed(val, mcbsp->st_data->io_base_st + reg);
}
static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg)
{
- return __raw_readl(mcbsp->st_data->io_base_st + reg);
+ return readl_relaxed(mcbsp->st_data->io_base_st + reg);
}
#define MCBSP_READ(mcbsp, reg) \
@@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \
unsigned long val; \
int status; \
\
- status = strict_strtoul(buf, 0, &val); \
+ status = kstrtoul(buf, 0, &val); \
if (status) \
return status; \
\
@@ -1012,25 +1012,32 @@ int omap_mcbsp_init(struct platform_device *pdev)
}
}
- res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx");
- if (!res) {
- dev_err(&pdev->dev, "invalid rx DMA channel\n");
- return -ENODEV;
- }
- /* RX DMA request number, and port address configuration */
- mcbsp->dma_data[1].name = "Audio Capture";
- mcbsp->dma_data[1].dma_req = res->start;
- mcbsp->dma_data[1].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 1);
+ if (!pdev->dev.of_node) {
+ res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx");
+ if (!res) {
+ dev_err(&pdev->dev, "invalid tx DMA channel\n");
+ return -ENODEV;
+ }
+ mcbsp->dma_req[0] = res->start;
+ mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0];
- res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx");
- if (!res) {
- dev_err(&pdev->dev, "invalid tx DMA channel\n");
- return -ENODEV;
+ res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx");
+ if (!res) {
+ dev_err(&pdev->dev, "invalid rx DMA channel\n");
+ return -ENODEV;
+ }
+ mcbsp->dma_req[1] = res->start;
+ mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1];
+ } else {
+ mcbsp->dma_data[0].filter_data = "tx";
+ mcbsp->dma_data[1].filter_data = "rx";
}
- /* TX DMA request number, and port address configuration */
- mcbsp->dma_data[0].name = "Audio Playback";
- mcbsp->dma_data[0].dma_req = res->start;
- mcbsp->dma_data[0].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 0);
+
+ mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0);
+ mcbsp->dma_data[0].maxburst = 4;
+
+ mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1);
+ mcbsp->dma_data[1].maxburst = 4;
mcbsp->fclk = clk_get(&pdev->dev, "fck");
if (IS_ERR(mcbsp->fclk)) {
diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h
index f93e0b0af30..96d1b086bcf 100644
--- a/sound/soc/omap/mcbsp.h
+++ b/sound/soc/omap/mcbsp.h
@@ -24,14 +24,14 @@
#ifndef __ASOC_MCBSP_H
#define __ASOC_MCBSP_H
-#include "omap-pcm.h"
-
#ifdef CONFIG_ARCH_OMAP1
#define mcbsp_omap1() 1
#else
#define mcbsp_omap1() 0
#endif
+#include <sound/dmaengine_pcm.h>
+
/* McBSP register numbers. Register address offset = num * reg_step */
enum {
/* Common registers */
@@ -312,7 +312,8 @@ struct omap_mcbsp {
struct omap_mcbsp_platform_data *pdata;
struct omap_mcbsp_st_data *st_data;
struct omap_mcbsp_reg_cfg cfg_regs;
- struct omap_pcm_dma_data dma_data[2];
+ struct snd_dmaengine_dai_dma_data dma_data[2];
+ unsigned int dma_req[2];
int dma_op_mode;
u16 max_tx_thres;
u16 max_rx_thres;
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 230b8c14484..5d7f9cebe04 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -34,7 +34,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define N810_HEADSET_AMP_GPIO 10
#define N810_SPEAKER_AMP_GPIO 101
@@ -69,26 +68,30 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm)
break;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (n810_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (line1l)
- snd_soc_dapm_enable_pin(dapm, "LINE1L");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINE1L");
else
- snd_soc_dapm_disable_pin(dapm, "LINE1L");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINE1L");
if (n810_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(dapm, "DMic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
+
+ snd_soc_dapm_sync_unlocked(dapm);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int n810_startup(struct snd_pcm_substream *substream)
@@ -101,12 +104,12 @@ static int n810_startup(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
n810_ext_control(&codec->dapm);
- return clk_enable(sys_clkout2);
+ return clk_prepare_enable(sys_clkout2);
}
static void n810_shutdown(struct snd_pcm_substream *substream)
{
- clk_disable(sys_clkout2);
+ clk_disable_unprepare(sys_clkout2);
}
static int n810_hw_params(struct snd_pcm_substream *substream,
@@ -230,8 +233,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Ext Spk", NULL, "LLOUT"},
{"Ext Spk", NULL, "RLOUT"},
- {"DMic Rate 64", NULL, "Mic Bias 2V"},
- {"Mic Bias 2V", NULL, "DMic"},
+ {"DMic Rate 64", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "DMic"},
};
static const char *spk_function[] = {"Off", "On"};
@@ -275,7 +278,7 @@ static struct snd_soc_dai_link n810_dai = {
.name = "TLV320AIC33",
.stream_name = "AIC33",
.cpu_dai_name = "omap-mcbsp.2",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.2",
.codec_name = "tlv320aic3x-codec.2-0018",
.codec_dai_name = "tlv320aic3x-hifi",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
@@ -306,7 +309,9 @@ static int __init n810_soc_init(void)
int err;
struct device *dev;
- if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
+ if (!of_have_populated_dt() ||
+ (!of_machine_is_compatible("nokia,n810") &&
+ !of_machine_is_compatible("nokia,n810-wimax")))
return -ENODEV;
n810_snd_device = platform_device_alloc("soc-audio", -1);
@@ -345,8 +350,11 @@ static int __init n810_soc_init(void)
clk_set_parent(sys_clkout2_src, func96m_clk);
clk_set_rate(sys_clkout2, 12000000);
- BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
- (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+ if (WARN_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+ (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0))) {
+ err = -EINVAL;
+ goto err4;
+ }
gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index e7d93fa412a..cec836ed0c0 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -23,7 +23,6 @@
#include <linux/clk.h>
#include <linux/platform_device.h>
#include <linux/mfd/twl6040.h>
-#include <linux/platform_data/omap-abe-twl6040.h>
#include <linux/module.h>
#include <linux/of.h>
@@ -34,7 +33,6 @@
#include "omap-dmic.h"
#include "omap-mcpdm.h"
-#include "omap-pcm.h"
#include "../codecs/twl6040.h"
struct abe_twl6040 {
@@ -49,8 +47,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int clk_id, freq;
int ret;
@@ -167,19 +164,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"AFMR", NULL, "Line In"},
};
-static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
- int connected, char *pin)
-{
- if (!connected)
- snd_soc_dapm_disable_pin(dapm, pin);
-}
-
static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_card *card = codec->card;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+ struct snd_soc_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
@@ -204,24 +192,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
}
- /*
- * NULL pdata means we booted with DT. In this case the routing is
- * provided and the card is fully routed, no need to mark pins.
- */
- if (!pdata)
- return ret;
-
- /* Disable not connected paths if not used */
- twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
- twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
- twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
- twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
- twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
- twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
- twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
-
return ret;
}
@@ -232,8 +202,7 @@ static const struct snd_soc_dapm_route dmic_audio_map[] = {
static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
@@ -244,9 +213,7 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
- .cpu_dai_name = "omap-mcpdm",
.codec_dai_name = "twl6040-legacy",
- .platform_name = "omap-pcm-audio",
.codec_name = "twl6040-codec",
.init = omap_abe_twl6040_init,
.ops = &omap_abe_ops,
@@ -254,9 +221,7 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
{
.name = "DMIC",
.stream_name = "DMIC Capture",
- .cpu_dai_name = "omap-dmic",
.codec_dai_name = "dmic-hifi",
- .platform_name = "omap-pcm-audio",
.codec_name = "dmic-codec",
.init = omap_abe_dmic_init,
.ops = &omap_abe_dmic_ops,
@@ -275,13 +240,18 @@ static struct snd_soc_card omap_abe_card = {
static int omap_abe_probe(struct platform_device *pdev)
{
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
struct device_node *node = pdev->dev.of_node;
struct snd_soc_card *card = &omap_abe_card;
+ struct device_node *dai_node;
struct abe_twl6040 *priv;
int num_links = 0;
int ret = 0;
+ if (!node) {
+ dev_err(&pdev->dev, "of node is missing.\n");
+ return -ENODEV;
+ }
+
card->dev = &pdev->dev;
priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
@@ -290,78 +260,50 @@ static int omap_abe_probe(struct platform_device *pdev)
priv->dmic_codec_dev = ERR_PTR(-EINVAL);
- if (node) {
- struct device_node *dai_node;
-
- if (snd_soc_of_parse_card_name(card, "ti,model")) {
- dev_err(&pdev->dev, "Card name is not provided\n");
- return -ENODEV;
- }
+ if (snd_soc_of_parse_card_name(card, "ti,model")) {
+ dev_err(&pdev->dev, "Card name is not provided\n");
+ return -ENODEV;
+ }
- ret = snd_soc_of_parse_audio_routing(card,
- "ti,audio-routing");
- if (ret) {
- dev_err(&pdev->dev,
- "Error while parsing DAPM routing\n");
- return ret;
- }
+ ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
+ if (ret) {
+ dev_err(&pdev->dev, "Error while parsing DAPM routing\n");
+ return ret;
+ }
- dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
- if (!dai_node) {
- dev_err(&pdev->dev, "McPDM node is not provided\n");
- return -EINVAL;
- }
- abe_twl6040_dai_links[0].cpu_dai_name = NULL;
- abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+ dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "McPDM node is not provided\n");
+ return -EINVAL;
+ }
+ abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+ abe_twl6040_dai_links[0].platform_of_node = dai_node;
- dai_node = of_parse_phandle(node, "ti,dmic", 0);
- if (dai_node) {
- num_links = 2;
- abe_twl6040_dai_links[1].cpu_dai_name = NULL;
- abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+ dai_node = of_parse_phandle(node, "ti,dmic", 0);
+ if (dai_node) {
+ num_links = 2;
+ abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+ abe_twl6040_dai_links[1].platform_of_node = dai_node;
- priv->dmic_codec_dev = platform_device_register_simple(
+ priv->dmic_codec_dev = platform_device_register_simple(
"dmic-codec", -1, NULL, 0);
- if (IS_ERR(priv->dmic_codec_dev)) {
- dev_err(&pdev->dev,
- "Can't instantiate dmic-codec\n");
- return PTR_ERR(priv->dmic_codec_dev);
- }
- } else {
- num_links = 1;
- }
-
- priv->jack_detection = of_property_read_bool(node,
- "ti,jack-detection");
- of_property_read_u32(node, "ti,mclk-freq",
- &priv->mclk_freq);
- if (!priv->mclk_freq) {
- dev_err(&pdev->dev, "MCLK frequency not provided\n");
- ret = -EINVAL;
- goto err_unregister;
+ if (IS_ERR(priv->dmic_codec_dev)) {
+ dev_err(&pdev->dev, "Can't instantiate dmic-codec\n");
+ return PTR_ERR(priv->dmic_codec_dev);
}
-
- omap_abe_card.fully_routed = 1;
- } else if (pdata) {
- if (pdata->card_name) {
- card->name = pdata->card_name;
- } else {
- dev_err(&pdev->dev, "Card name is not provided\n");
- return -ENODEV;
- }
-
- if (pdata->has_dmic)
- num_links = 2;
- else
- num_links = 1;
-
- priv->jack_detection = pdata->jack_detection;
- priv->mclk_freq = pdata->mclk_freq;
} else {
- dev_err(&pdev->dev, "Missing pdata\n");
- return -ENODEV;
+ num_links = 1;
+ }
+
+ priv->jack_detection = of_property_read_bool(node, "ti,jack-detection");
+ of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
+ if (!priv->mclk_freq) {
+ dev_err(&pdev->dev, "MCLK frequency not provided\n");
+ ret = -EINVAL;
+ goto err_unregister;
}
+ card->fully_routed = 1;
if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index ba49ccd9eed..6925d714121 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -39,8 +39,9 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
-#include "omap-pcm.h"
#include "omap-dmic.h"
struct omap_dmic {
@@ -55,23 +56,18 @@ struct omap_dmic {
u32 ch_enabled;
bool active;
struct mutex mutex;
-};
-/*
- * Stream DMA parameters
- */
-static struct omap_pcm_dma_data omap_dmic_dai_dma_params = {
- .name = "DMIC capture",
+ struct snd_dmaengine_dai_dma_data dma_data;
};
static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val)
{
- __raw_writel(val, dmic->io_base + reg);
+ writel_relaxed(val, dmic->io_base + reg);
}
static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg)
{
- return __raw_readl(dmic->io_base + reg);
+ return readl_relaxed(dmic->io_base + reg);
}
static inline void omap_dmic_start(struct omap_dmic *dmic)
@@ -118,7 +114,6 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
mutex_unlock(&dmic->mutex);
- snd_soc_dai_set_dma_data(dai, substream, &omap_dmic_dai_dma_params);
return ret;
}
@@ -203,7 +198,7 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
int channels;
dmic->clk_div = omap_dmic_select_divider(dmic, params_rate(params));
@@ -230,7 +225,7 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
/* packet size is threshold * channels */
dma_data = snd_soc_dai_get_dma_data(dai, substream);
- dma_data->packet_size = dmic->threshold * channels;
+ dma_data->maxburst = dmic->threshold * channels;
return 0;
}
@@ -422,6 +417,9 @@ static int omap_dmic_probe(struct snd_soc_dai *dai)
/* Configure DMIC threshold value */
dmic->threshold = OMAP_DMIC_THRES_MAX - 3;
+
+ snd_soc_dai_init_dma_data(dai, NULL, &dmic->dma_data);
+
return 0;
}
@@ -448,6 +446,10 @@ static struct snd_soc_dai_driver omap_dmic_dai = {
.ops = &omap_dmic_dai_ops,
};
+static const struct snd_soc_component_driver omap_dmic_component = {
+ .name = "omap-dmic",
+};
+
static int asoc_dmic_probe(struct platform_device *pdev)
{
struct omap_dmic *dmic;
@@ -476,38 +478,24 @@ static int asoc_dmic_probe(struct platform_device *pdev)
ret = -ENODEV;
goto err_put_clk;
}
- omap_dmic_dai_dma_params.port_addr = res->start + OMAP_DMIC_DATA_REG;
+ dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG;
- res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
- if (!res) {
- dev_err(dmic->dev, "invalid dma resource\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
- omap_dmic_dai_dma_params.dma_req = res->start;
+ dmic->dma_data.filter_data = "up_link";
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
- if (!res) {
- dev_err(dmic->dev, "invalid memory resource\n");
- ret = -ENODEV;
+ dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(dmic->io_base)) {
+ ret = PTR_ERR(dmic->io_base);
goto err_put_clk;
}
- if (!devm_request_mem_region(&pdev->dev, res->start,
- resource_size(res), pdev->name)) {
- dev_err(dmic->dev, "memory region already claimed\n");
- ret = -ENODEV;
- goto err_put_clk;
- }
- dmic->io_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!dmic->io_base) {
- ret = -ENOMEM;
+ ret = snd_soc_register_component(&pdev->dev, &omap_dmic_component,
+ &omap_dmic_dai, 1);
+ if (ret)
goto err_put_clk;
- }
- ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai);
+ ret = omap_pcm_platform_register(&pdev->dev);
if (ret)
goto err_put_clk;
@@ -522,7 +510,7 @@ static int asoc_dmic_remove(struct platform_device *pdev)
{
struct omap_dmic *dmic = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
clk_put(dmic->fclk);
return 0;
diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c
index d4eaa92e518..f649fe84b62 100644
--- a/sound/soc/omap/omap-hdmi-card.c
+++ b/sound/soc/omap/omap-hdmi-card.c
@@ -33,9 +33,9 @@ static struct snd_soc_dai_link omap_hdmi_dai = {
.name = "HDMI",
.stream_name = "HDMI",
.cpu_dai_name = "omap-hdmi-audio-dai",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-hdmi-audio-dai",
.codec_name = "hdmi-audio-codec",
- .codec_dai_name = "omap-hdmi-hifi",
+ .codec_dai_name = "hdmi-hifi",
};
static struct snd_soc_card snd_soc_omap_hdmi = {
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index 7ea24819d57..eb9c39299f8 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -32,15 +32,17 @@
#include <sound/soc.h>
#include <sound/asound.h>
#include <sound/asoundef.h>
+#include <sound/dmaengine_pcm.h>
#include <video/omapdss.h>
+#include <sound/omap-pcm.h>
-#include "omap-pcm.h"
#include "omap-hdmi.h"
#define DRV_NAME "omap-hdmi-audio-dai"
struct hdmi_priv {
- struct omap_pcm_dma_data dma_params;
+ struct snd_dmaengine_dai_dma_data dma_data;
+ unsigned int dma_req;
struct omap_dss_audio dss_audio;
struct snd_aes_iec958 iec;
struct snd_cea_861_aud_if cea;
@@ -68,7 +70,7 @@ static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
return -ENODEV;
}
- snd_soc_dai_set_dma_data(dai, substream, &priv->dma_params);
+ snd_soc_dai_set_dma_data(dai, substream, &priv->dma_data);
return 0;
}
@@ -88,28 +90,25 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
struct snd_aes_iec958 *iec = &priv->iec;
struct snd_cea_861_aud_if *cea = &priv->cea;
- struct omap_pcm_dma_data *dma_data;
int err = 0;
- dma_data = snd_soc_dai_get_dma_data(dai, substream);
-
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- dma_data->packet_size = 16;
+ priv->dma_data.maxburst = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- dma_data->packet_size = 32;
+ priv->dma_data.maxburst = 32;
break;
default:
dev_err(dai->dev, "format not supported!\n");
return -EINVAL;
}
- dma_data->data_type = 32;
-
/*
* fill the IEC-60958 channel status word
*/
+ /* initialize the word bytes */
+ memset(iec->status, 0, sizeof(iec->status));
/* specify IEC-60958-3 (commercial use) */
iec->status[0] &= ~IEC958_AES0_PROFESSIONAL;
@@ -262,6 +261,10 @@ static struct snd_soc_dai_driver omap_hdmi_dai = {
.ops = &omap_hdmi_dai_ops,
};
+static const struct snd_soc_component_driver omap_hdmi_component = {
+ .name = DRV_NAME,
+};
+
static int omap_hdmi_probe(struct platform_device *pdev)
{
int ret;
@@ -281,8 +284,7 @@ static int omap_hdmi_probe(struct platform_device *pdev)
return -ENODEV;
}
- hdmi_data->dma_params.port_addr = hdmi_rsrc->start
- + OMAP_HDMI_AUDIO_DMA_PORT;
+ hdmi_data->dma_data.addr = hdmi_rsrc->start + OMAP_HDMI_AUDIO_DMA_PORT;
hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!hdmi_rsrc) {
@@ -290,8 +292,9 @@ static int omap_hdmi_probe(struct platform_device *pdev)
return -ENODEV;
}
- hdmi_data->dma_params.dma_req = hdmi_rsrc->start;
- hdmi_data->dma_params.name = "HDMI playback";
+ hdmi_data->dma_req = hdmi_rsrc->start;
+ hdmi_data->dma_data.filter_data = &hdmi_data->dma_req;
+ hdmi_data->dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
/*
* TODO: We assume that there is only one DSS HDMI device. Future
@@ -319,16 +322,20 @@ static int omap_hdmi_probe(struct platform_device *pdev)
}
dev_set_drvdata(&pdev->dev, hdmi_data);
- ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai);
+ ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component,
+ &omap_hdmi_dai, 1);
+
+ if (ret)
+ return ret;
- return ret;
+ return omap_pcm_platform_register(&pdev->dev);
}
static int omap_hdmi_remove(struct platform_device *pdev)
{
struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
if (hdmi_data == NULL) {
dev_err(&pdev->dev, "cannot obtain HDMi data\n");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8d2defd6fdb..efe2cd699b7 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -33,11 +33,12 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "mcbsp.h"
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000)
@@ -62,24 +63,22 @@ enum {
* Stream DMA parameters. DMA request line and port address are set runtime
* since they are different between OMAP1 and later OMAPs
*/
-static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
+ unsigned int packet_size)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
- struct omap_pcm_dma_data *dma_data;
int words;
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
/*
* Configure McBSP threshold based on either:
* packet_size, when the sDMA is in packet mode, or based on the
* period size in THRESHOLD mode, otherwise use McBSP threshold = 1
* for mono streams.
*/
- if (dma_data->packet_size)
- words = dma_data->packet_size;
+ if (packet_size)
+ words = packet_size;
else
words = 1;
@@ -151,9 +150,6 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
}
- snd_soc_dai_set_dma_data(cpu_dai, substream,
- &mcbsp->dma_data[substream->stream]);
-
return err;
}
@@ -226,7 +222,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
{
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
int wlen, channels, wpf;
int pkt_size = 0;
unsigned int format, div, framesize, master;
@@ -245,7 +241,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
if (mcbsp->pdata->buffer_size) {
- dma_data->set_threshold = omap_mcbsp_set_threshold;
if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
int period_words, max_thrsh;
int divider = 0;
@@ -276,9 +271,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
/* Use packet mode for non mono streams */
pkt_size = channels;
}
+ omap_mcbsp_set_threshold(substream, pkt_size);
}
- dma_data->packet_size = pkt_size;
+ dma_data->maxburst = pkt_size;
if (mcbsp->configured) {
/* McBSP already configured by another stream */
@@ -435,6 +431,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
/* Sample rate generator drives the FS */
regs->srgr2 |= FSGM;
break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ /* McBSP slave. FS clock as output */
+ regs->srgr2 |= FSGM;
+ regs->pcr0 |= FSXM;
+ break;
case SND_SOC_DAIFMT_CBM_CFM:
/* McBSP slave */
break;
@@ -556,6 +557,10 @@ static int omap_mcbsp_probe(struct snd_soc_dai *dai)
pm_runtime_enable(mcbsp->dev);
+ snd_soc_dai_init_dma_data(dai,
+ &mcbsp->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+ &mcbsp->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
return 0;
}
@@ -586,6 +591,10 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = {
.ops = &mcbsp_dai_ops,
};
+static const struct snd_soc_component_driver omap_mcbsp_component = {
+ .name = "omap-mcbsp",
+};
+
static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -684,7 +693,7 @@ OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \
OMAP_MCBSP_ST_CONTROLS(2);
OMAP_MCBSP_ST_CONTROLS(3);
-int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
+int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id)
{
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
@@ -694,7 +703,7 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
return 0;
}
- switch (mcbsp->id) {
+ switch (port_id) {
case 2: /* McBSP 2 */
return snd_soc_add_dai_controls(cpu_dai,
omap_mcbsp2_st_controls,
@@ -704,6 +713,7 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd)
omap_mcbsp3_st_controls,
ARRAY_SIZE(omap_mcbsp3_st_controls));
default:
+ dev_err(mcbsp->dev, "Port %d not supported\n", port_id);
break;
}
@@ -792,17 +802,22 @@ static int asoc_mcbsp_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, mcbsp);
ret = omap_mcbsp_init(pdev);
- if (!ret)
- return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai);
+ if (ret)
+ return ret;
- return ret;
+ ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component,
+ &omap_mcbsp_dai, 1);
+ if (ret)
+ return ret;
+
+ return omap_pcm_platform_register(&pdev->dev);
}
static int asoc_mcbsp_remove(struct platform_device *pdev)
{
struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
if (mcbsp->pdata->ops && mcbsp->pdata->ops->free)
mcbsp->pdata->ops->free(mcbsp->id);
@@ -811,8 +826,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev)
clk_put(mcbsp->fclk);
- platform_set_drvdata(pdev, NULL);
-
return 0;
}
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index ba8386a0d8d..2e3369c27be 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -39,6 +39,6 @@ enum omap_mcbsp_div {
OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */
};
-int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd);
+int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id);
#endif
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 2fe8be20945..f0e2ebeab02 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -39,11 +39,15 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
#include "omap-mcpdm.h"
-#include "omap-pcm.h"
-#define OMAP44XX_MCPDM_L3_BASE 0x49032000
+struct mcpdm_link_config {
+ u32 link_mask; /* channel mask for the direction */
+ u32 threshold; /* FIFO threshold */
+};
struct omap_mcpdm {
struct device *dev;
@@ -53,38 +57,30 @@ struct omap_mcpdm {
struct mutex mutex;
- /* channel data */
- u32 dn_channels;
- u32 up_channels;
-
- /* McPDM FIFO thresholds */
- u32 dn_threshold;
- u32 up_threshold;
+ /* Playback/Capture configuration */
+ struct mcpdm_link_config config[2];
/* McPDM dn offsets for rx1, and 2 channels */
u32 dn_rx_offset;
+
+ /* McPDM needs to be restarted due to runtime reconfiguration */
+ bool restart;
+
+ struct snd_dmaengine_dai_dma_data dma_data[2];
};
/*
* Stream DMA parameters
*/
-static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = {
- {
- .name = "Audio playback",
- },
- {
- .name = "Audio capture",
- },
-};
static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val)
{
- __raw_writel(val, mcpdm->io_base + reg);
+ writel_relaxed(val, mcpdm->io_base + reg);
}
static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg)
{
- return __raw_readl(mcpdm->io_base + reg);
+ return readl_relaxed(mcpdm->io_base + reg);
}
#ifdef DEBUG
@@ -130,11 +126,12 @@ static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {}
static void omap_mcpdm_start(struct omap_mcpdm *mcpdm)
{
u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+ u32 link_mask = mcpdm->config[0].link_mask | mcpdm->config[1].link_mask;
ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
- ctrl |= mcpdm->dn_channels | mcpdm->up_channels;
+ ctrl |= link_mask;
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
@@ -148,11 +145,12 @@ static void omap_mcpdm_start(struct omap_mcpdm *mcpdm)
static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm)
{
u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+ u32 link_mask = MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK;
ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
- ctrl &= ~(mcpdm->dn_channels | mcpdm->up_channels);
+ ctrl &= ~(link_mask);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
@@ -188,8 +186,10 @@ static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm)
omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset);
}
- omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN, mcpdm->dn_threshold);
- omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP, mcpdm->up_threshold);
+ omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN,
+ mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold);
+ omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP,
+ mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold);
omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET,
MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE);
@@ -266,9 +266,6 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
}
mutex_unlock(&mcpdm->mutex);
- snd_soc_dai_set_dma_data(dai, substream,
- &omap_mcpdm_dai_dma_params[substream->stream]);
-
return 0;
}
@@ -283,6 +280,8 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
if (omap_mcpdm_active(mcpdm)) {
omap_mcpdm_stop(mcpdm);
omap_mcpdm_close_streams(mcpdm);
+ mcpdm->config[0].link_mask = 0;
+ mcpdm->config[1].link_mask = 0;
}
}
@@ -295,7 +294,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
int stream = substream->stream;
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
+ u32 threshold;
int channels;
int link_mask = 0;
@@ -325,16 +325,32 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
dma_data = snd_soc_dai_get_dma_data(dai, substream);
+ threshold = mcpdm->config[stream].threshold;
/* Configure McPDM channels, and DMA packet size */
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
- mcpdm->dn_channels = link_mask << 3;
- dma_data->packet_size =
- (MCPDM_DN_THRES_MAX - mcpdm->dn_threshold) * channels;
+ link_mask <<= 3;
+
+ /* If capture is not running assume a stereo stream to come */
+ if (!mcpdm->config[!stream].link_mask)
+ mcpdm->config[!stream].link_mask = 0x3;
+
+ dma_data->maxburst =
+ (MCPDM_DN_THRES_MAX - threshold) * channels;
} else {
- mcpdm->up_channels = link_mask << 0;
- dma_data->packet_size = mcpdm->up_threshold * channels;
+ /* If playback is not running assume a stereo stream to come */
+ if (!mcpdm->config[!stream].link_mask)
+ mcpdm->config[!stream].link_mask = (0x3 << 3);
+
+ dma_data->maxburst = threshold * channels;
}
+ /* Check if we need to restart McPDM with this stream */
+ if (mcpdm->config[stream].link_mask &&
+ mcpdm->config[stream].link_mask != link_mask)
+ mcpdm->restart = true;
+
+ mcpdm->config[stream].link_mask = link_mask;
+
return 0;
}
@@ -346,6 +362,11 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
if (!omap_mcpdm_active(mcpdm)) {
omap_mcpdm_start(mcpdm);
omap_mcpdm_reg_dump(mcpdm);
+ } else if (mcpdm->restart) {
+ omap_mcpdm_stop(mcpdm);
+ omap_mcpdm_start(mcpdm);
+ mcpdm->restart = false;
+ omap_mcpdm_reg_dump(mcpdm);
}
return 0;
@@ -369,7 +390,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
pm_runtime_get_sync(mcpdm->dev);
omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00);
- ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+ ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler,
0, "McPDM", (void *)mcpdm);
pm_runtime_put_sync(mcpdm->dev);
@@ -380,8 +401,14 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
}
/* Configure McPDM threshold values */
- mcpdm->dn_threshold = 2;
- mcpdm->up_threshold = MCPDM_UP_THRES_MAX - 3;
+ mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold = 2;
+ mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold =
+ MCPDM_UP_THRES_MAX - 3;
+
+ snd_soc_dai_init_dma_data(dai,
+ &mcpdm->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+ &mcpdm->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
return ret;
}
@@ -389,7 +416,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
{
struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
- free_irq(mcpdm->irq, (void *)mcpdm);
pm_runtime_disable(mcpdm->dev);
return 0;
@@ -420,6 +446,10 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = {
.ops = &omap_mcpdm_dai_ops,
};
+static const struct snd_soc_component_driver omap_mcpdm_component = {
+ .name = "omap-mcpdm",
+};
+
void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
u8 rx1, u8 rx2)
{
@@ -433,6 +463,7 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
{
struct omap_mcpdm *mcpdm;
struct resource *res;
+ int ret;
mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL);
if (!mcpdm)
@@ -446,37 +477,16 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
if (res == NULL)
return -ENOMEM;
- omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA;
- omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA;
-
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (res == NULL)
- return -ENOMEM;
-
- res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link");
- if (!res)
- return -ENODEV;
-
- omap_mcpdm_dai_dma_params[0].dma_req = res->start;
-
- res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link");
- if (!res)
- return -ENODEV;
+ mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA;
+ mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA;
- omap_mcpdm_dai_dma_params[1].dma_req = res->start;
+ mcpdm->dma_data[0].filter_data = "dn_link";
+ mcpdm->dma_data[1].filter_data = "up_link";
res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
- if (res == NULL)
- return -ENOMEM;
-
- if (!devm_request_mem_region(&pdev->dev, res->start,
- resource_size(res), "McPDM"))
- return -EBUSY;
-
- mcpdm->io_base = devm_ioremap(&pdev->dev, res->start,
- resource_size(res));
- if (!mcpdm->io_base)
- return -ENOMEM;
+ mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(mcpdm->io_base))
+ return PTR_ERR(mcpdm->io_base);
mcpdm->irq = platform_get_irq(pdev, 0);
if (mcpdm->irq < 0)
@@ -484,13 +494,13 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
mcpdm->dev = &pdev->dev;
- return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai);
-}
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &omap_mcpdm_component,
+ &omap_mcpdm_dai, 1);
+ if (ret)
+ return ret;
-static int asoc_mcpdm_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_dai(&pdev->dev);
- return 0;
+ return omap_pcm_platform_register(&pdev->dev);
}
static const struct of_device_id omap_mcpdm_of_match[] = {
@@ -507,7 +517,6 @@ static struct platform_driver asoc_mcpdm_driver = {
},
.probe = asoc_mcpdm_probe,
- .remove = asoc_mcpdm_remove,
};
module_platform_driver(asoc_mcpdm_driver);
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 47bdbd415ad..8d809f8509c 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -32,8 +32,6 @@
#include <sound/dmaengine_pcm.h>
#include <sound/soc.h>
-#include "omap-pcm.h"
-
#ifdef CONFIG_ARCH_OMAP1
#define pcm_omap1510() cpu_is_omap1510()
#else
@@ -47,8 +45,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
- .formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FMTBIT_S32_LE,
.period_bytes_min = 32,
.period_bytes_max = 64 * 1024,
.periods_min = 2,
@@ -56,25 +52,6 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
.buffer_bytes_max = 128 * 1024,
};
-static int omap_pcm_get_dma_buswidth(int num_bits)
-{
- int buswidth;
-
- switch (num_bits) {
- case 16:
- buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
- break;
- case 32:
- buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
- break;
- default:
- buswidth = -EINVAL;
- break;
- }
- return buswidth;
-}
-
-
/* this may get called several times by oss emulation */
static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -105,20 +82,9 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
if (err)
return err;
- /* Override the *_dma addr_width if requested by the DAI driver */
- if (dma_data->data_type) {
- int buswidth = omap_pcm_get_dma_buswidth(dma_data->data_type);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- config.dst_addr_width = buswidth;
- else
- config.src_addr_width = buswidth;
- }
-
- config.src_addr = dma_data->port_addr;
- config.dst_addr = dma_data->port_addr;
- config.src_maxburst = dma_data->packet_size;
- config.dst_maxburst = dma_data->packet_size;
+ snd_dmaengine_pcm_set_config_from_dai_data(substream,
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+ &config);
return dmaengine_slave_config(chan, &config);
}
@@ -129,37 +95,6 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
return 0;
}
-static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct omap_pcm_dma_data *dma_data;
- int ret = 0;
-
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_RESUME:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- /* Configure McBSP internal buffer usage */
- if (dma_data->set_threshold)
- dma_data->set_threshold(substream);
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- break;
- default:
- ret = -EINVAL;
- }
-
- if (ret == 0)
- ret = snd_dmaengine_pcm_trigger(substream, cmd);
-
- return ret;
-}
-
static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
{
snd_pcm_uframes_t offset;
@@ -174,29 +109,27 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
static int omap_pcm_open(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct omap_pcm_dma_data *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
int ret;
snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
- /* Ensure that buffer size is a multiple of period size */
- ret = snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS);
- if (ret < 0)
- return ret;
-
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- ret = snd_dmaengine_pcm_open(substream, omap_dma_filter_fn,
- &dma_data->dma_req);
- return ret;
-}
-static int omap_pcm_close(struct snd_pcm_substream *substream)
-{
- snd_dmaengine_pcm_close(substream);
- return 0;
+ /* DT boot: filter_data is the DMA name */
+ if (rtd->cpu_dai->dev->of_node) {
+ struct dma_chan *chan;
+
+ chan = dma_request_slave_channel(rtd->cpu_dai->dev,
+ dma_data->filter_data);
+ ret = snd_dmaengine_pcm_open(substream, chan);
+ } else {
+ ret = snd_dmaengine_pcm_open_request_chan(substream,
+ omap_dma_filter_fn,
+ dma_data->filter_data);
+ }
+ return ret;
}
static int omap_pcm_mmap(struct snd_pcm_substream *substream,
@@ -212,17 +145,15 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream,
static struct snd_pcm_ops omap_pcm_ops = {
.open = omap_pcm_open,
- .close = omap_pcm_close,
+ .close = snd_dmaengine_pcm_close_release_chan,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = omap_pcm_hw_params,
.hw_free = omap_pcm_hw_free,
- .trigger = omap_pcm_trigger,
+ .trigger = snd_dmaengine_pcm_trigger,
.pointer = omap_pcm_pointer,
.mmap = omap_pcm_mmap,
};
-static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
-
static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
int stream)
{
@@ -267,12 +198,11 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
struct snd_pcm *pcm = rtd->pcm;
- int ret = 0;
+ int ret;
- 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(64);
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+ if (ret)
+ return ret;
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = omap_pcm_preallocate_dma_buffer(pcm,
@@ -302,31 +232,12 @@ static struct snd_soc_platform_driver omap_soc_platform = {
.pcm_free = omap_pcm_free_dma_buffers,
};
-static int omap_pcm_probe(struct platform_device *pdev)
-{
- return snd_soc_register_platform(&pdev->dev,
- &omap_soc_platform);
-}
-
-static int omap_pcm_remove(struct platform_device *pdev)
+int omap_pcm_platform_register(struct device *dev)
{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
+ return devm_snd_soc_register_platform(dev, &omap_soc_platform);
}
-
-static struct platform_driver omap_pcm_driver = {
- .driver = {
- .name = "omap-pcm-audio",
- .owner = THIS_MODULE,
- },
-
- .probe = omap_pcm_probe,
- .remove = omap_pcm_remove,
-};
-
-module_platform_driver(omap_pcm_driver);
+EXPORT_SYMBOL_GPL(omap_pcm_platform_register);
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:omap-pcm-audio");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
deleted file mode 100644
index cabe74c4068..00000000000
--- a/sound/soc/omap/omap-pcm.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * omap-pcm.h
- *
- * Copyright (C) 2008 Nokia Corporation
- *
- * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
- * Peter Ujfalusi <peter.ujfalusi@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
- *
- */
-
-#ifndef __OMAP_PCM_H__
-#define __OMAP_PCM_H__
-
-struct snd_pcm_substream;
-
-struct omap_pcm_dma_data {
- char *name; /* stream identifier */
- int dma_req; /* DMA request line */
- unsigned long port_addr; /* transmit/receive register */
- void (*set_threshold)(struct snd_pcm_substream *substream);
- int data_type; /* 8, 16, 32 (bits) or 0 to let omap-pcm
- * to decide the sDMA data type */
- int packet_size; /* packet size only in PACKET mode */
-};
-
-#endif
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index 4541d28b531..f8a6adc2d81 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -11,6 +11,8 @@
* omap3evm (Author: Anuj Aggarwal <anuj.aggarwal@ti.com>)
* overo (Author: Steve Sakoman <steve@sakoman.com>)
* igep0020 (Author: Enric Balletbo i Serra <eballetbo@iseebcn.com>)
+ * zoom2 (Author: Misael Lopez Cruz <misael.lopez@ti.com>)
+ * sdp3430 (Author: Misael Lopez Cruz <misael.lopez@ti.com>)
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -32,13 +34,20 @@
#include <linux/platform_data/omap-twl4030.h>
#include <linux/module.h>
#include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
+#include <sound/jack.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
+
+struct omap_twl4030 {
+ int jack_detect; /* board can detect jack events */
+ struct snd_soc_jack hs_jack;
+};
static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
@@ -46,8 +55,7 @@ static int omap_twl4030_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_card *card = codec->card;
+ struct snd_soc_card *card = rtd->card;
unsigned int fmt;
int ret;
@@ -87,24 +95,189 @@ static struct snd_soc_ops omap_twl4030_ops = {
.hw_params = omap_twl4030_hw_params,
};
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Earpiece Spk", NULL),
+ SND_SOC_DAPM_SPK("Handsfree Spk", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_SPK("Carkit Spk", NULL),
+
+ SND_SOC_DAPM_MIC("Main Mic", NULL),
+ SND_SOC_DAPM_MIC("Sub Mic", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Carkit Mic", NULL),
+ SND_SOC_DAPM_MIC("Digital0 Mic", NULL),
+ SND_SOC_DAPM_MIC("Digital1 Mic", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Headset Stereophone: HSOL, HSOR */
+ {"Headset Stereophone", NULL, "HSOL"},
+ {"Headset Stereophone", NULL, "HSOR"},
+ /* External Speakers: HFL, HFR */
+ {"Handsfree Spk", NULL, "HFL"},
+ {"Handsfree Spk", NULL, "HFR"},
+ /* External Speakers: PredrivL, PredrivR */
+ {"Ext Spk", NULL, "PREDRIVEL"},
+ {"Ext Spk", NULL, "PREDRIVER"},
+ /* Carkit speakers: CARKITL, CARKITR */
+ {"Carkit Spk", NULL, "CARKITL"},
+ {"Carkit Spk", NULL, "CARKITR"},
+ /* Earpiece */
+ {"Earpiece Spk", NULL, "EARPIECE"},
+
+ /* External Mics: MAINMIC, SUBMIC with bias */
+ {"MAINMIC", NULL, "Main Mic"},
+ {"Main Mic", NULL, "Mic Bias 1"},
+ {"SUBMIC", NULL, "Sub Mic"},
+ {"Sub Mic", NULL, "Mic Bias 2"},
+ /* Headset Mic: HSMIC with bias */
+ {"HSMIC", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "Headset Mic Bias"},
+ /* Digital Mics: DIGIMIC0, DIGIMIC1 with bias */
+ {"DIGIMIC0", NULL, "Digital0 Mic"},
+ {"Digital0 Mic", NULL, "Mic Bias 1"},
+ {"DIGIMIC1", NULL, "Digital1 Mic"},
+ {"Digital1 Mic", NULL, "Mic Bias 2"},
+ /* Carkit In: CARKITMIC */
+ {"CARKITMIC", NULL, "Carkit Mic"},
+ /* Aux In: AUXL, AUXR */
+ {"AUXL", NULL, "Line In"},
+ {"AUXR", NULL, "Line In"},
+};
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headset Stereophone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+ {
+ .name = "hsdet-gpio",
+ .report = SND_JACK_HEADSET,
+ .debounce_time = 200,
+ },
+};
+
+static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm,
+ int connected, char *pin)
+{
+ if (!connected)
+ snd_soc_dapm_disable_pin(dapm, pin);
+}
+
+static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_card *card = rtd->card;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev);
+ struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card);
+ int ret = 0;
+
+ /* Headset jack detection only if it is supported */
+ if (priv->jack_detect > 0) {
+ hs_jack_gpios[0].gpio = priv->jack_detect;
+
+ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
+ &priv->hs_jack);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_pins(&priv->hs_jack,
+ ARRAY_SIZE(hs_jack_pins),
+ hs_jack_pins);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_jack_add_gpios(&priv->hs_jack,
+ ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+ if (ret)
+ return ret;
+ }
+
+ /*
+ * NULL pdata means we booted with DT. In this case the routing is
+ * provided and the card is fully routed, no need to mark pins.
+ */
+ if (!pdata || !pdata->custom_routing)
+ return ret;
+
+ /* Disable not connected paths if not used */
+ twl4030_disconnect_pin(dapm, pdata->has_ear, "Earpiece Spk");
+ twl4030_disconnect_pin(dapm, pdata->has_hf, "Handsfree Spk");
+ twl4030_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
+ twl4030_disconnect_pin(dapm, pdata->has_predriv, "Ext Spk");
+ twl4030_disconnect_pin(dapm, pdata->has_carkit, "Carkit Spk");
+
+ twl4030_disconnect_pin(dapm, pdata->has_mainmic, "Main Mic");
+ twl4030_disconnect_pin(dapm, pdata->has_submic, "Sub Mic");
+ twl4030_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
+ twl4030_disconnect_pin(dapm, pdata->has_carkitmic, "Carkit Mic");
+ twl4030_disconnect_pin(dapm, pdata->has_digimic0, "Digital0 Mic");
+ twl4030_disconnect_pin(dapm, pdata->has_digimic1, "Digital1 Mic");
+ twl4030_disconnect_pin(dapm, pdata->has_linein, "Line In");
+
+ return ret;
+}
+
+static int omap_twl4030_card_remove(struct snd_soc_card *card)
+{
+ struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card);
+
+ if (priv->jack_detect > 0)
+ snd_soc_jack_free_gpios(&priv->hs_jack,
+ ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+
+ return 0;
+}
+
/* Digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
{
- .name = "TWL4030",
- .stream_name = "TWL4030",
+ .name = "TWL4030 HiFi",
+ .stream_name = "TWL4030 HiFi",
.cpu_dai_name = "omap-mcbsp.2",
.codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.2",
.codec_name = "twl4030-codec",
+ .init = omap_twl4030_init,
.ops = &omap_twl4030_ops,
},
+ {
+ .name = "TWL4030 Voice",
+ .stream_name = "TWL4030 Voice",
+ .cpu_dai_name = "omap-mcbsp.3",
+ .codec_dai_name = "twl4030-voice",
+ .platform_name = "omap-mcbsp.2",
+ .codec_name = "twl4030-codec",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ },
};
/* Audio machine driver */
static struct snd_soc_card omap_twl4030_card = {
.owner = THIS_MODULE,
+ .remove = omap_twl4030_card_remove,
.dai_link = omap_twl4030_dai_links,
.num_links = ARRAY_SIZE(omap_twl4030_dai_links),
+
+ .dapm_widgets = dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static int omap_twl4030_probe(struct platform_device *pdev)
@@ -112,12 +285,18 @@ static int omap_twl4030_probe(struct platform_device *pdev)
struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev);
struct device_node *node = pdev->dev.of_node;
struct snd_soc_card *card = &omap_twl4030_card;
+ struct omap_twl4030 *priv;
int ret = 0;
card->dev = &pdev->dev;
+ priv = devm_kzalloc(&pdev->dev, sizeof(struct omap_twl4030), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
if (node) {
struct device_node *dai_node;
+ struct property *prop;
if (snd_soc_of_parse_card_name(card, "ti,model")) {
dev_err(&pdev->dev, "Card name is not provided\n");
@@ -132,6 +311,33 @@ static int omap_twl4030_probe(struct platform_device *pdev)
omap_twl4030_dai_links[0].cpu_dai_name = NULL;
omap_twl4030_dai_links[0].cpu_of_node = dai_node;
+ omap_twl4030_dai_links[0].platform_name = NULL;
+ omap_twl4030_dai_links[0].platform_of_node = dai_node;
+
+ dai_node = of_parse_phandle(node, "ti,mcbsp-voice", 0);
+ if (!dai_node) {
+ card->num_links = 1;
+ } else {
+ omap_twl4030_dai_links[1].cpu_dai_name = NULL;
+ omap_twl4030_dai_links[1].cpu_of_node = dai_node;
+
+ omap_twl4030_dai_links[1].platform_name = NULL;
+ omap_twl4030_dai_links[1].platform_of_node = dai_node;
+ }
+
+ priv->jack_detect = of_get_named_gpio(node,
+ "ti,jack-det-gpio", 0);
+
+ /* Optional: audio routing can be provided */
+ prop = of_find_property(node, "ti,audio-routing", NULL);
+ if (prop) {
+ ret = snd_soc_of_parse_audio_routing(card,
+ "ti,audio-routing");
+ if (ret)
+ return ret;
+
+ card->fully_routed = 1;
+ }
} else if (pdata) {
if (pdata->card_name) {
card->name = pdata->card_name;
@@ -139,14 +345,20 @@ static int omap_twl4030_probe(struct platform_device *pdev)
dev_err(&pdev->dev, "Card name is not provided\n");
return -ENODEV;
}
+
+ if (!pdata->voice_connected)
+ card->num_links = 1;
+
+ priv->jack_detect = pdata->jack_detect;
} else {
dev_err(&pdev->dev, "Missing pdata\n");
return -ENODEV;
}
- ret = snd_soc_register_card(card);
+ snd_soc_card_set_drvdata(card, priv);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret) {
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
ret);
return ret;
}
@@ -154,15 +366,6 @@ static int omap_twl4030_probe(struct platform_device *pdev)
return 0;
}
-static int omap_twl4030_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
-}
-
static const struct of_device_id omap_twl4030_of_match[] = {
{.compatible = "ti,omap-twl4030", },
{ },
@@ -177,7 +380,6 @@ static struct platform_driver omap_twl4030_driver = {
.of_match_table = omap_twl4030_of_match,
},
.probe = omap_twl4030_probe,
- .remove = omap_twl4030_remove,
};
module_platform_driver(omap_twl4030_driver);
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 43d950a79ff..076bec606d7 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -34,7 +34,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#define OMAP3_PANDORA_DAC_POWER_GPIO 118
#define OMAP3_PANDORA_AMP_POWER_GPIO 14
@@ -80,12 +79,18 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
+ int ret;
+
/*
* The PCM1773 DAC datasheet requires 1ms delay between switching
* VCC power on/off and /PD pin high/low
*/
if (SND_SOC_DAPM_EVENT_ON(event)) {
- regulator_enable(omap3pandora_dac_reg);
+ ret = regulator_enable(omap3pandora_dac_reg);
+ if (ret) {
+ dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret);
+ return ret;
+ }
mdelay(1);
gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
} else {
@@ -116,7 +121,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
* |A| <~~clk~~+
* |P| <--- TWL4030 <--------- Line In and MICs
*/
-static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget omap3pandora_dapm_widgets[] = {
SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM,
0, 0, omap3pandora_dac_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -125,37 +130,32 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
-};
-static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Mic (internal)", NULL),
SND_SOC_DAPM_MIC("Mic (external)", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
};
-static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+static const struct snd_soc_dapm_route omap3pandora_map[] = {
{"PCM DAC", NULL, "APLL Enable"},
{"Headphone Amplifier", NULL, "PCM DAC"},
{"Line Out", NULL, "PCM DAC"},
{"Headphone Jack", NULL, "Headphone Amplifier"},
-};
-static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
{"AUXL", NULL, "Line In"},
{"AUXR", NULL, "Line In"},
- {"MAINMIC", NULL, "Mic Bias 1"},
- {"Mic Bias 1", NULL, "Mic (internal)"},
+ {"MAINMIC", NULL, "Mic (internal)"},
+ {"Mic (internal)", NULL, "Mic Bias 1"},
- {"SUBMIC", NULL, "Mic Bias 2"},
- {"Mic Bias 2", NULL, "Mic (external)"},
+ {"SUBMIC", NULL, "Mic (external)"},
+ {"Mic (external)", NULL, "Mic Bias 2"},
};
static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
/* All TWL4030 output pins are floating */
snd_soc_dapm_nc_pin(dapm, "EARPIECE");
@@ -169,20 +169,13 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "HFR");
snd_soc_dapm_nc_pin(dapm, "VIBRA");
- ret = snd_soc_dapm_new_controls(dapm, omap3pandora_out_dapm_widgets,
- ARRAY_SIZE(omap3pandora_out_dapm_widgets));
- if (ret < 0)
- return ret;
-
- return snd_soc_dapm_add_routes(dapm, omap3pandora_out_map,
- ARRAY_SIZE(omap3pandora_out_map));
+ return 0;
}
static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
/* Not comnnected */
snd_soc_dapm_nc_pin(dapm, "HSMIC");
@@ -190,13 +183,7 @@ static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
- ret = snd_soc_dapm_new_controls(dapm, omap3pandora_in_dapm_widgets,
- ARRAY_SIZE(omap3pandora_in_dapm_widgets));
- if (ret < 0)
- return ret;
-
- return snd_soc_dapm_add_routes(dapm, omap3pandora_in_map,
- ARRAY_SIZE(omap3pandora_in_map));
+ return 0;
}
static struct snd_soc_ops omap3pandora_ops = {
@@ -210,7 +197,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
.stream_name = "HiFi Out",
.cpu_dai_name = "omap-mcbsp.2",
.codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.2",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
@@ -221,7 +208,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
.stream_name = "Line/Mic In",
.cpu_dai_name = "omap-mcbsp.4",
.codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.4",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
@@ -236,6 +223,11 @@ static struct snd_soc_card snd_soc_card_omap3pandora = {
.owner = THIS_MODULE,
.dai_link = omap3pandora_dai,
.num_links = ARRAY_SIZE(omap3pandora_dai),
+
+ .dapm_widgets = omap3pandora_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(omap3pandora_dapm_widgets),
+ .dapm_routes = omap3pandora_map,
+ .num_dapm_routes = ARRAY_SIZE(omap3pandora_map),
};
static struct platform_device *omap3pandora_snd_device;
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 06ef8d67ed1..aa4053bf671 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -33,7 +33,6 @@
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
#include "../codecs/tlv320aic23.h"
#define CODEC_CLOCK 12000000
@@ -97,7 +96,7 @@ static struct snd_soc_dai_link osk_dai = {
.stream_name = "AIC23",
.cpu_dai_name = "omap-mcbsp.1",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.1",
.codec_name = "tlv320aic23-codec",
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index d921ddbe3ec..943922c79f7 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -26,6 +26,7 @@
#include <linux/delay.h>
#include <linux/gpio.h>
#include <linux/platform_device.h>
+#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/jack.h>
@@ -37,16 +38,6 @@
#include <asm/mach-types.h>
#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-#define RX51_TVOUT_SEL_GPIO 40
-#define RX51_JACK_DETECT_GPIO 177
-#define RX51_ECI_SW_GPIO 182
-/*
- * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
- * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
- */
-#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7)
enum {
RX51_JACK_DISABLED,
@@ -55,12 +46,21 @@ enum {
RX51_JACK_HS, /* headset: stereo output with mic */
};
+struct rx51_audio_pdata {
+ struct gpio_desc *tvout_selection_gpio;
+ struct gpio_desc *jack_detection_gpio;
+ struct gpio_desc *eci_sw_gpio;
+ struct gpio_desc *speaker_amp_gpio;
+};
+
static int rx51_spk_func;
static int rx51_dmic_func;
static int rx51_jack_func;
static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
{
+ struct snd_soc_card *card = dapm->card;
+ struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
int hp = 0, hs = 0, tvout = 0;
switch (rx51_jack_func) {
@@ -75,26 +75,30 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
break;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (rx51_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (rx51_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(dapm, "DMic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (hs)
- snd_soc_dapm_enable_pin(dapm, "HS Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
else
- snd_soc_dapm_disable_pin(dapm, "HS Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
+
+ gpiod_set_value(pdata->tvout_selection_gpio, tvout);
- gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
+ snd_soc_dapm_sync_unlocked(dapm);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int rx51_startup(struct snd_pcm_substream *substream)
@@ -151,10 +155,12 @@ static int rx51_set_spk(struct snd_kcontrol *kcontrol,
static int rx51_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
- if (SND_SOC_DAPM_EVENT_ON(event))
- gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 1);
- else
- gpio_set_value_cansleep(RX51_SPEAKER_AMP_TWL_GPIO, 0);
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+
+ gpiod_set_raw_value_cansleep(pdata->speaker_amp_gpio,
+ !!SND_SOC_DAPM_EVENT_ON(event));
return 0;
}
@@ -220,7 +226,6 @@ static struct snd_soc_jack rx51_av_jack;
static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
{
- .gpio = RX51_JACK_DETECT_GPIO,
.name = "avdet-gpio",
.report = SND_JACK_HEADSET,
.invert = 1,
@@ -234,9 +239,6 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event),
SND_SOC_DAPM_MIC("HS Mic", NULL),
SND_SOC_DAPM_LINE("FM Transmitter", NULL),
-};
-
-static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = {
SND_SOC_DAPM_SPK("Earphone", NULL),
};
@@ -248,21 +250,21 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"FM Transmitter", NULL, "LLOUT"},
{"FM Transmitter", NULL, "RLOUT"},
- {"DMic Rate 64", NULL, "Mic Bias 2V"},
- {"Mic Bias 2V", NULL, "DMic"},
-};
+ {"DMic Rate 64", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "DMic"},
-static const struct snd_soc_dapm_route audio_mapb[] = {
{"b LINE2R", NULL, "MONO_LOUT"},
{"Earphone", NULL, "b HPLOUT"},
- {"LINE1L", NULL, "b Mic Bias 2.5V"},
- {"b Mic Bias 2.5V", NULL, "HS Mic"}
+ {"LINE1L", NULL, "b Mic Bias"},
+ {"b Mic Bias", NULL, "HS Mic"}
};
-static const char *spk_function[] = {"Off", "On"};
-static const char *input_function[] = {"ADC", "Digital Mic"};
-static const char *jack_function[] = {"Off", "TV-OUT", "Headphone", "Headset"};
+static const char * const spk_function[] = {"Off", "On"};
+static const char * const input_function[] = {"ADC", "Digital Mic"};
+static const char * const jack_function[] = {
+ "Off", "TV-OUT", "Headphone", "Headset"
+};
static const struct soc_enum rx51_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
@@ -278,15 +280,15 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
SOC_ENUM_EXT("Jack Function", rx51_enum[2],
rx51_get_jack, rx51_set_jack),
SOC_DAPM_PIN_SWITCH("FM Transmitter"),
-};
-
-static const struct snd_kcontrol_new aic34_rx51_controlsb[] = {
SOC_DAPM_PIN_SWITCH("Earphone"),
};
static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_card *card = rtd->card;
+ struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+
struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
@@ -295,57 +297,49 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "MIC3R");
snd_soc_dapm_nc_pin(dapm, "LINE1R");
- /* Add RX-51 specific controls */
- err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls,
- ARRAY_SIZE(aic34_rx51_controls));
- if (err < 0)
- return err;
-
- /* Add RX-51 specific widgets */
- snd_soc_dapm_new_controls(dapm, aic34_dapm_widgets,
- ARRAY_SIZE(aic34_dapm_widgets));
-
- /* Set up RX-51 specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
err = tpa6130a2_add_controls(codec);
- if (err < 0)
+ if (err < 0) {
+ dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
return err;
+ }
snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
- err = omap_mcbsp_st_add_controls(rtd);
- if (err < 0)
+ err = omap_mcbsp_st_add_controls(rtd, 2);
+ if (err < 0) {
+ dev_err(card->dev, "Failed to add MCBSP controls\n");
return err;
+ }
/* AV jack detection */
err = snd_soc_jack_new(codec, "AV Jack",
SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
&rx51_av_jack);
- if (err)
+ if (err) {
+ dev_err(card->dev, "Failed to add AV Jack\n");
return err;
+ }
+
+ /* prepare gpio for snd_soc_jack_add_gpios */
+ rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio);
+ devm_gpiod_put(card->dev, pdata->jack_detection_gpio);
+
err = snd_soc_jack_add_gpios(&rx51_av_jack,
ARRAY_SIZE(rx51_av_jack_gpios),
rx51_av_jack_gpios);
+ if (err) {
+ dev_err(card->dev, "Failed to add GPIOs\n");
+ return err;
+ }
return err;
}
-static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
+static int rx51_card_remove(struct snd_soc_card *card)
{
- int err;
-
- err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb,
- ARRAY_SIZE(aic34_rx51_controlsb));
- if (err < 0)
- return err;
-
- err = snd_soc_dapm_new_controls(dapm, aic34_dapm_widgetsb,
- ARRAY_SIZE(aic34_dapm_widgetsb));
- if (err < 0)
- return 0;
+ snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
+ rx51_av_jack_gpios);
- return snd_soc_dapm_add_routes(dapm, audio_mapb,
- ARRAY_SIZE(audio_mapb));
+ return 0;
}
/* Digital audio interface glue - connects codec <--> CPU */
@@ -355,7 +349,7 @@ static struct snd_soc_dai_link rx51_dai[] = {
.stream_name = "AIC34",
.cpu_dai_name = "omap-mcbsp.2",
.codec_dai_name = "tlv320aic3x-hifi",
- .platform_name = "omap-pcm-audio",
+ .platform_name = "omap-mcbsp.2",
.codec_name = "tlv320aic3x-codec.2-0018",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,
@@ -368,7 +362,6 @@ static struct snd_soc_aux_dev rx51_aux_dev[] = {
{
.name = "TLV320AIC34b",
.codec_name = "tlv320aic3x-codec.2-0019",
- .init = rx51_aic34b_init,
},
};
@@ -383,69 +376,158 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = {
static struct snd_soc_card rx51_sound_card = {
.name = "RX-51",
.owner = THIS_MODULE,
+ .remove = rx51_card_remove,
.dai_link = rx51_dai,
.num_links = ARRAY_SIZE(rx51_dai),
.aux_dev = rx51_aux_dev,
.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
.codec_conf = rx51_codec_conf,
.num_configs = ARRAY_SIZE(rx51_codec_conf),
-};
-static struct platform_device *rx51_snd_device;
+ .controls = aic34_rx51_controls,
+ .num_controls = ARRAY_SIZE(aic34_rx51_controls),
+ .dapm_widgets = aic34_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic34_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
-static int __init rx51_soc_init(void)
+static int rx51_soc_probe(struct platform_device *pdev)
{
+ struct rx51_audio_pdata *pdata;
+ struct device_node *np = pdev->dev.of_node;
+ struct snd_soc_card *card = &rx51_sound_card;
int err;
- if (!machine_is_nokia_rx51())
+ if (!machine_is_nokia_rx51() && !of_machine_is_compatible("nokia,omap3-n900"))
return -ENODEV;
- err = gpio_request_one(RX51_TVOUT_SEL_GPIO,
- GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel");
- if (err)
- goto err_gpio_tvout_sel;
- err = gpio_request_one(RX51_ECI_SW_GPIO,
- GPIOF_DIR_OUT | GPIOF_INIT_HIGH, "eci_sw");
- if (err)
- goto err_gpio_eci_sw;
-
- rx51_snd_device = platform_device_alloc("soc-audio", -1);
- if (!rx51_snd_device) {
- err = -ENOMEM;
- goto err1;
+ card->dev = &pdev->dev;
+
+ if (np) {
+ struct device_node *dai_node;
+
+ dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "McBSP node is not provided\n");
+ return -EINVAL;
+ }
+ rx51_dai[0].cpu_dai_name = NULL;
+ rx51_dai[0].platform_name = NULL;
+ rx51_dai[0].cpu_of_node = dai_node;
+ rx51_dai[0].platform_of_node = dai_node;
+
+ dai_node = of_parse_phandle(np, "nokia,audio-codec", 0);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "Codec node is not provided\n");
+ return -EINVAL;
+ }
+ rx51_dai[0].codec_name = NULL;
+ rx51_dai[0].codec_of_node = dai_node;
+
+ dai_node = of_parse_phandle(np, "nokia,audio-codec", 1);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n");
+ return -EINVAL;
+ }
+ rx51_aux_dev[0].codec_name = NULL;
+ rx51_aux_dev[0].codec_of_node = dai_node;
+ rx51_codec_conf[0].dev_name = NULL;
+ rx51_codec_conf[0].of_node = dai_node;
+
+ dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0);
+ if (!dai_node) {
+ dev_err(&pdev->dev, "Headphone amplifier node is not provided\n");
+ return -EINVAL;
+ }
+
+ /* TODO: tpa6130a2a driver supports only a single instance, so
+ * this driver ignores the headphone-amplifier node for now.
+ * It's already mandatory in the DT binding to be future proof.
+ */
}
- platform_set_drvdata(rx51_snd_device, &rx51_sound_card);
+ pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+ if (pdata == NULL) {
+ dev_err(card->dev, "failed to create private data\n");
+ return -ENOMEM;
+ }
+ snd_soc_card_set_drvdata(card, pdata);
- err = platform_device_add(rx51_snd_device);
- if (err)
- goto err2;
+ pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,
+ "tvout-selection");
+ if (IS_ERR(pdata->tvout_selection_gpio)) {
+ dev_err(card->dev, "could not get tvout selection gpio\n");
+ return PTR_ERR(pdata->tvout_selection_gpio);
+ }
- return 0;
-err2:
- platform_device_put(rx51_snd_device);
-err1:
- gpio_free(RX51_ECI_SW_GPIO);
-err_gpio_eci_sw:
- gpio_free(RX51_TVOUT_SEL_GPIO);
-err_gpio_tvout_sel:
+ err = gpiod_direction_output(pdata->tvout_selection_gpio, 0);
+ if (err) {
+ dev_err(card->dev, "could not setup tvout selection gpio\n");
+ return err;
+ }
- return err;
-}
+ pdata->jack_detection_gpio = devm_gpiod_get(card->dev,
+ "jack-detection");
+ if (IS_ERR(pdata->jack_detection_gpio)) {
+ dev_err(card->dev, "could not get jack detection gpio\n");
+ return PTR_ERR(pdata->jack_detection_gpio);
+ }
-static void __exit rx51_soc_exit(void)
-{
- snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
- rx51_av_jack_gpios);
+ pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch");
+ if (IS_ERR(pdata->eci_sw_gpio)) {
+ dev_err(card->dev, "could not get eci switch gpio\n");
+ return PTR_ERR(pdata->eci_sw_gpio);
+ }
- platform_device_unregister(rx51_snd_device);
- gpio_free(RX51_ECI_SW_GPIO);
- gpio_free(RX51_TVOUT_SEL_GPIO);
+ err = gpiod_direction_output(pdata->eci_sw_gpio, 1);
+ if (err) {
+ dev_err(card->dev, "could not setup eci switch gpio\n");
+ return err;
+ }
+
+ pdata->speaker_amp_gpio = devm_gpiod_get(card->dev,
+ "speaker-amplifier");
+ if (IS_ERR(pdata->speaker_amp_gpio)) {
+ dev_err(card->dev, "could not get speaker enable gpio\n");
+ return PTR_ERR(pdata->speaker_amp_gpio);
+ }
+
+ err = gpiod_direction_output(pdata->speaker_amp_gpio, 0);
+ if (err) {
+ dev_err(card->dev, "could not setup speaker enable gpio\n");
+ return err;
+ }
+
+ err = devm_snd_soc_register_card(card->dev, card);
+ if (err) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err);
+ return err;
+ }
+
+ return 0;
}
-module_init(rx51_soc_init);
-module_exit(rx51_soc_exit);
+#if defined(CONFIG_OF)
+static const struct of_device_id rx51_audio_of_match[] = {
+ { .compatible = "nokia,n900-audio", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, rx51_audio_of_match);
+#endif
+
+static struct platform_driver rx51_soc_driver = {
+ .driver = {
+ .name = "rx51-audio",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(rx51_audio_of_match),
+ },
+ .probe = rx51_soc_probe,
+};
+
+module_platform_driver(rx51_soc_driver);
MODULE_AUTHOR("Nokia Corporation");
MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:rx51-audio");
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
deleted file mode 100644
index b462a2c9385..00000000000
--- a/sound/soc/omap/sdp3430.c
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- * sdp3430.c -- SoC audio for TI OMAP3430 SDP
- *
- * Author: Misael Lopez Cruz <x0052729@ti.com>
- *
- * Based on:
- * Author: Steve Sakoman <steve@sakoman.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 <linux/i2c/twl.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/gpio-omap.h>
-#include <linux/platform_data/asoc-ti-mcbsp.h>
-
-/* Register descriptions for twl4030 codec part */
-#include <linux/mfd/twl4030-audio.h>
-#include <linux/module.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.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,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- int 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_ops = {
- .hw_params = sdp3430_hw_params,
-};
-
-/* Headset jack */
-static struct snd_soc_jack hs_jack;
-
-/* Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- {
- .pin = "Headset Mic",
- .mask = SND_JACK_MICROPHONE,
- },
- {
- .pin = "Headset Stereophone",
- .mask = SND_JACK_HEADPHONE,
- },
-};
-
-/* Headset jack detection gpios */
-static struct snd_soc_jack_gpio hs_jack_gpios[] = {
- {
- .gpio = (OMAP_MAX_GPIO_LINES + 2),
- .name = "hsdet-gpio",
- .report = SND_JACK_HEADSET,
- .debounce_time = 200,
- },
-};
-
-/* SDP3430 machine DAPM */
-static const struct snd_soc_dapm_widget sdp3430_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),
-};
-
-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 Mic: HSMIC with bias */
- {"HSMIC", NULL, "Headset Mic Bias"},
- {"Headset Mic Bias", NULL, "Headset Mic"},
-
- /* Headset Stereophone (Headphone): HSOL, HSOR */
- {"Headset Stereophone", NULL, "HSOL"},
- {"Headset Stereophone", NULL, "HSOR"},
-};
-
-static int sdp3430_twl4030_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- /* SDP3430 connected pins */
- snd_soc_dapm_enable_pin(dapm, "Ext Mic");
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
-
- /* TWL4030 not connected pins */
- snd_soc_dapm_nc_pin(dapm, "AUXL");
- snd_soc_dapm_nc_pin(dapm, "AUXR");
- snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
-
- snd_soc_dapm_nc_pin(dapm, "OUTL");
- snd_soc_dapm_nc_pin(dapm, "OUTR");
- snd_soc_dapm_nc_pin(dapm, "EARPIECE");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
- snd_soc_dapm_nc_pin(dapm, "CARKITL");
- snd_soc_dapm_nc_pin(dapm, "CARKITR");
-
- /* Headset jack detection */
- ret = snd_soc_jack_new(codec, "Headset Jack",
- SND_JACK_HEADSET, &hs_jack);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
- hs_jack_pins);
- if (ret)
- return ret;
-
- ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
-
- return ret;
-}
-
-static int sdp3430_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- unsigned short reg;
-
- /* Enable voice interface */
- reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
- reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
- codec->driver->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 I2S",
- .stream_name = "TWL4030 Audio",
- .cpu_dai_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = sdp3430_twl4030_init,
- .ops = &sdp3430_ops,
- },
- {
- .name = "TWL4030 PCM",
- .stream_name = "TWL4030 Voice",
- .cpu_dai_name = "omap-mcbsp.3",
- .codec_dai_name = "twl4030-voice",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = sdp3430_twl4030_voice_init,
- .ops = &sdp3430_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_sdp3430 = {
- .name = "SDP3430",
- .owner = THIS_MODULE,
- .dai_link = sdp3430_dai,
- .num_links = ARRAY_SIZE(sdp3430_dai),
-
- .dapm_widgets = sdp3430_twl4030_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(sdp3430_twl4030_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *sdp3430_snd_device;
-
-static int __init sdp3430_soc_init(void)
-{
- int ret;
- u8 pin_mux;
-
- if (!machine_is_omap_3430sdp())
- return -ENODEV;
- printk(KERN_INFO "SDP3430 SoC init\n");
-
- sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
- if (!sdp3430_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(sdp3430_snd_device, &snd_soc_sdp3430);
-
- /* Set TWL4030 GPIO6 as EXTMUTE signal */
- twl_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);
- twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
- TWL4030_INTBR_PMBR1);
-
- ret = platform_device_add(sdp3430_snd_device);
- if (ret)
- goto err1;
-
- return 0;
-
-err1:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(sdp3430_snd_device);
-
- return ret;
-}
-module_init(sdp3430_soc_init);
-
-static void __exit sdp3430_soc_exit(void)
-{
- snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
-
- platform_device_unregister(sdp3430_snd_device);
-}
-module_exit(sdp3430_soc_exit);
-
-MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC SDP3430");
-MODULE_LICENSE("GPL");
-
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
deleted file mode 100644
index 771bff27ac3..00000000000
--- a/sound/soc/omap/zoom2.c
+++ /dev/null
@@ -1,207 +0,0 @@
-/*
- * 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 <linux/gpio.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-ti-mcbsp.h>
-#include <linux/platform_data/gpio-omap.h>
-
-/* Register descriptions for twl4030 codec part */
-#include <linux/mfd/twl4030-audio.h>
-#include <linux/module.h>
-
-#include "omap-mcbsp.h"
-#include "omap-pcm.h"
-
-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->codec_dai;
- int 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,
-};
-
-/* 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_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* TWL4030 not connected pins */
- snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
- snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
- snd_soc_dapm_nc_pin(dapm, "EARPIECE");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
- snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
- snd_soc_dapm_nc_pin(dapm, "CARKITL");
- snd_soc_dapm_nc_pin(dapm, "CARKITR");
-
- return 0;
-}
-
-static int zoom2_twl4030_voice_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- unsigned short reg;
-
- /* Enable voice interface */
- reg = codec->driver->read(codec, TWL4030_REG_VOICE_IF);
- reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
- codec->driver->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_name = "omap-mcbsp.2",
- .codec_dai_name = "twl4030-hifi",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = zoom2_twl4030_init,
- .ops = &zoom2_ops,
- },
- {
- .name = "TWL4030 PCM",
- .stream_name = "TWL4030 Voice",
- .cpu_dai_name = "omap-mcbsp.3",
- .codec_dai_name = "twl4030-voice",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl4030-codec",
- .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = zoom2_twl4030_voice_init,
- .ops = &zoom2_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_zoom2 = {
- .name = "Zoom2",
- .owner = THIS_MODULE,
- .dai_link = zoom2_dai,
- .num_links = ARRAY_SIZE(zoom2_dai),
-
- .dapm_widgets = zoom2_twl4030_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(zoom2_twl4030_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *zoom2_snd_device;
-
-static int __init zoom2_soc_init(void)
-{
- int ret;
-
- if (!machine_is_omap_zoom2())
- 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, &snd_soc_zoom2);
- ret = platform_device_add(zoom2_snd_device);
- if (ret)
- goto err1;
-
- 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)
-{
- 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 v1.2.3-70-g09d2 (git 2.46.0) at 2025-11-09 17:09:10 +0000