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-rw-r--r--sound/soc/omap/Kconfig4
-rw-r--r--sound/soc/omap/am3517evm.c2
-rw-r--r--sound/soc/omap/ams-delta.c80
-rw-r--r--sound/soc/omap/n810.c2
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c13
-rw-r--r--sound/soc/omap/omap-dmic.c9
-rw-r--r--sound/soc/omap/omap-hdmi-card.c2
-rw-r--r--sound/soc/omap/omap-hdmi.c6
-rw-r--r--sound/soc/omap/omap-mcbsp.c25
-rw-r--r--sound/soc/omap/omap-mcbsp.h2
-rw-r--r--sound/soc/omap/omap-mcpdm.c16
-rw-r--r--sound/soc/omap/omap-pcm.c25
-rw-r--r--sound/soc/omap/omap-twl4030.c42
-rw-r--r--sound/soc/omap/omap3pandora.c35
-rw-r--r--sound/soc/omap/osk5912.c2
-rw-r--r--sound/soc/omap/rx51.c275
16 files changed, 297 insertions, 243 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index e00659351a4..d44463a7b0f 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810
config SND_OMAP_SOC_RX51
tristate "SoC Audio support for Nokia RX-51"
- depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST)
+ 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
@@ -37,7 +37,7 @@ config SND_OMAP_SOC_RX51
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
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 994dcf34597..25a33e9d417 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -77,7 +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 56a5219c0a0..0cc41f94de4 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -38,7 +38,6 @@
#include "omap-mcbsp.h"
#include "../codecs/cx20442.h"
-
/* Board specific DAPM widgets */
static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
/* Handset */
@@ -90,17 +89,23 @@ 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->items)
@@ -166,8 +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") <<
@@ -270,12 +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)
{
@@ -302,7 +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);
@@ -475,15 +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) {
@@ -520,40 +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;
}
@@ -565,7 +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,
};
@@ -574,8 +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 */
@@ -603,10 +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/n810.c b/sound/soc/omap/n810.c
index fd4d9c809e5..5d7f9cebe04 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -278,7 +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 |
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 024dafc3e29..cec836ed0c0 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -47,8 +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;
@@ -168,7 +167,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
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_card *card = rtd->card;
struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
@@ -214,9 +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,
@@ -224,9 +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,
@@ -281,14 +276,14 @@ static int omap_abe_probe(struct platform_device *pdev)
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;
+ 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;
+ abe_twl6040_dai_links[1].platform_of_node = dai_node;
priv->dmic_codec_dev = platform_device_register_simple(
"dmic-codec", -1, NULL, 0);
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 1bd531d718f..6925d714121 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -40,6 +40,7 @@
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
#include "omap-dmic.h"
@@ -113,7 +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, &dmic->dma_data);
return ret;
}
@@ -417,6 +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;
}
@@ -492,6 +495,10 @@ static int asoc_dmic_probe(struct platform_device *pdev)
if (ret)
goto err_put_clk;
+ ret = omap_pcm_platform_register(&pdev->dev);
+ if (ret)
+ goto err_put_clk;
+
return 0;
err_put_clk:
diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c
index 7e66e9cba5a..f649fe84b62 100644
--- a/sound/soc/omap/omap-hdmi-card.c
+++ b/sound/soc/omap/omap-hdmi-card.c
@@ -33,7 +33,7 @@ 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 = "hdmi-hifi",
};
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index ced3b88b44d..eb9c39299f8 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -34,6 +34,7 @@
#include <sound/asoundef.h>
#include <sound/dmaengine_pcm.h>
#include <video/omapdss.h>
+#include <sound/omap-pcm.h>
#include "omap-hdmi.h"
@@ -324,7 +325,10 @@ static int omap_hdmi_probe(struct platform_device *pdev)
ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component,
&omap_hdmi_dai, 1);
- return ret;
+ if (ret)
+ return ret;
+
+ return omap_pcm_platform_register(&pdev->dev);
}
static int omap_hdmi_remove(struct platform_device *pdev)
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6c19bba2357..efe2cd699b7 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -34,6 +34,7 @@
#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"
@@ -149,9 +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;
}
@@ -559,6 +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;
}
@@ -691,7 +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);
@@ -701,7 +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,
@@ -711,6 +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;
}
@@ -799,11 +802,15 @@ 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_component(&pdev->dev, &omap_mcbsp_component,
- &omap_mcbsp_dai, 1);
+ if (ret)
+ return ret;
+
+ ret = snd_soc_register_component(&pdev->dev, &omap_mcbsp_component,
+ &omap_mcbsp_dai, 1);
+ if (ret)
+ return ret;
- return ret;
+ return omap_pcm_platform_register(&pdev->dev);
}
static int asoc_mcbsp_remove(struct platform_device *pdev)
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 2f5b1536477..f0e2ebeab02 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -40,6 +40,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
#include "omap-mcpdm.h"
@@ -265,9 +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,
- &mcpdm->dma_data[substream->stream]);
-
return 0;
}
@@ -406,6 +404,11 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
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;
}
@@ -460,6 +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)
@@ -490,9 +494,13 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
mcpdm->dev = &pdev->dev;
- return devm_snd_soc_register_component(&pdev->dev,
+ ret = devm_snd_soc_register_component(&pdev->dev,
&omap_mcpdm_component,
&omap_mcpdm_dai, 1);
+ if (ret)
+ return ret;
+
+ return omap_pcm_platform_register(&pdev->dev);
}
static const struct of_device_id omap_mcpdm_of_match[] = {
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 07b8b7bc9d2..8d809f8509c 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -232,31 +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)
+int omap_pcm_platform_register(struct device *dev)
{
- return snd_soc_register_platform(&pdev->dev,
- &omap_soc_platform);
+ return devm_snd_soc_register_platform(dev, &omap_soc_platform);
}
-
-static int omap_pcm_remove(struct platform_device *pdev)
-{
- snd_soc_unregister_platform(&pdev->dev);
- return 0;
-}
-
-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-twl4030.c b/sound/soc/omap/omap-twl4030.c
index 6a8d6b5f160..f8a6adc2d81 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -55,8 +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;
@@ -179,7 +178,7 @@ static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm,
static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_card *card = codec->card;
+ 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);
@@ -232,6 +231,18 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
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[] = {
{
@@ -239,7 +250,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.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,
@@ -249,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp.3",
.codec_dai_name = "twl4030-voice",
- .platform_name = "omap-pcm-audio",
+ .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,
@@ -259,6 +270,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
/* 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),
@@ -299,12 +311,18 @@ 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,
@@ -348,19 +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);
- 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;
-}
-
static const struct of_device_id omap_twl4030_of_match[] = {
{.compatible = "ti,omap-twl4030", },
{ },
@@ -375,7 +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 cf604a2faa1..076bec606d7 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -121,7 +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),
@@ -130,22 +130,18 @@ 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"},
@@ -160,7 +156,6 @@ 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");
@@ -174,20 +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");
@@ -195,13 +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 = {
@@ -215,7 +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,
@@ -226,7 +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,
@@ -241,6 +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 d03e57da770..aa4053bf671 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -96,7 +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 7fb3d4b1037..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>
@@ -38,15 +39,6 @@
#include "omap-mcbsp.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,
RX51_JACK_TVOUT, /* tv-out with stereo output */
@@ -54,12 +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) {
@@ -93,7 +94,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
else
snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
- gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
+ gpiod_set_value(pdata->tvout_selection_gpio, tvout);
snd_soc_dapm_sync_unlocked(dapm);
@@ -154,10 +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;
}
@@ -223,7 +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,
@@ -237,9 +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),
};
@@ -253,9 +252,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"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"},
@@ -263,9 +260,11 @@ static const struct snd_soc_dapm_route audio_mapb[] = {
{"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),
@@ -281,15 +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;
@@ -298,57 +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 */
@@ -358,7 +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,
@@ -371,7 +362,6 @@ static struct snd_soc_aux_dev rx51_aux_dev[] = {
{
.name = "TLV320AIC34b",
.codec_name = "tlv320aic3x-codec.2-0019",
- .init = rx51_aic34b_init,
},
};
@@ -386,69 +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() && !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");
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-18 10:14:18 +0000