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-rw-r--r--sound/soc/omap/Kconfig13
-rw-r--r--sound/soc/omap/Makefile4
-rw-r--r--sound/soc/omap/ams-delta.c2
-rw-r--r--sound/soc/omap/n810.c17
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c349
-rw-r--r--sound/soc/omap/omap-dmic.c7
-rw-r--r--sound/soc/omap/omap-mcbsp.c8
-rw-r--r--sound/soc/omap/omap-mcbsp.h2
-rw-r--r--sound/soc/omap/omap-mcpdm.c2
-rw-r--r--sound/soc/omap/rx51.c25
-rw-r--r--sound/soc/omap/sdp4430.c279
11 files changed, 390 insertions, 318 deletions
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index fb1bf2581ef..47b23fea20c 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -97,16 +97,19 @@ config SND_OMAP_SOC_SDP3430
Say Y if you want to add support for SoC audio on Texas Instruments
SDP3430.
-config SND_OMAP_SOC_SDP4430
- tristate "SoC Audio support for Texas Instruments SDP4430"
- depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP
+config SND_OMAP_SOC_OMAP_ABE_TWL6040
+ tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
+ depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4
select SND_OMAP_SOC_DMIC
select SND_OMAP_SOC_MCPDM
select SND_SOC_TWL6040
select SND_SOC_DMIC
help
- Say Y if you want to add support for SoC audio on Texas Instruments
- SDP4430.
+ Say Y if you want to add support for SoC audio on OMAP boards using
+ ABE and twl6040 codec. This driver currently supports:
+ - SDP4430/Blaze boards
+ - PandaBoard (4430)
+ - PandaBoardES (4460)
config SND_OMAP_SOC_OMAP4_HDMI
tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 1fd723fb559..123ac18303e 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -20,7 +20,7 @@ snd-soc-overo-objs := overo.o
snd-soc-omap3evm-objs := omap3evm.o
snd-soc-am3517evm-objs := am3517evm.o
snd-soc-sdp3430-objs := sdp3430.o
-snd-soc-sdp4430-objs := sdp4430.o
+snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
snd-soc-zoom2-objs := zoom2.o
@@ -36,7 +36,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
-obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index f610260065b..41586b26ce9 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -544,7 +544,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_disable_pin(dapm, "AGCOUT");
/* Add virtual switch */
- ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+ ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls,
ARRAY_SIZE(ams_delta_audio_controls));
if (ret)
dev_warn(card->dev,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 597be412f1e..c292bf0fd19 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -55,9 +55,8 @@ static int n810_spk_func;
static int n810_jack_func;
static int n810_dmic_func;
-static void n810_ext_control(struct snd_soc_codec *codec)
+static void n810_ext_control(struct snd_soc_dapm_context *dapm)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
int hp = 0, line1l = 0;
switch (n810_jack_func) {
@@ -102,7 +101,7 @@ static int n810_startup(struct snd_pcm_substream *substream)
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
- n810_ext_control(codec);
+ n810_ext_control(&codec->dapm);
return clk_enable(sys_clkout2);
}
@@ -142,13 +141,13 @@ static int n810_get_spk(struct snd_kcontrol *kcontrol,
static int n810_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_spk_func == ucontrol->value.integer.value[0])
return 0;
n810_spk_func = ucontrol->value.integer.value[0];
- n810_ext_control(codec);
+ n810_ext_control(&card->dapm);
return 1;
}
@@ -164,13 +163,13 @@ static int n810_get_jack(struct snd_kcontrol *kcontrol,
static int n810_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_jack_func == ucontrol->value.integer.value[0])
return 0;
n810_jack_func = ucontrol->value.integer.value[0];
- n810_ext_control(codec);
+ n810_ext_control(&card->dapm);
return 1;
}
@@ -186,13 +185,13 @@ static int n810_get_input(struct snd_kcontrol *kcontrol,
static int n810_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_dmic_func == ucontrol->value.integer.value[0])
return 0;
n810_dmic_func = ucontrol->value.integer.value[0];
- n810_ext_control(codec);
+ n810_ext_control(&card->dapm);
return 1;
}
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
new file mode 100644
index 00000000000..93bb8eee22b
--- /dev/null
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -0,0 +1,349 @@
+/*
+ * omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and
+ * twl6040 codec
+ *
+ * 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
+ * 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/mfd/twl6040.h>
+#include <linux/platform_data/omap-abe-twl6040.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+
+#include "omap-dmic.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+
+static int omap_abe_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;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_card *card = codec->card;
+ struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+ int clk_id, freq;
+ int ret;
+
+ clk_id = twl6040_get_clk_id(rtd->codec);
+ if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
+ freq = pdata->mclk_freq;
+ else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
+ freq = 32768;
+ else
+ return -EINVAL;
+
+ /* set the codec mclk */
+ ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+ return ret;
+}
+
+static struct snd_soc_ops omap_abe_ops = {
+ .hw_params = omap_abe_hw_params,
+};
+
+static int omap_abe_dmic_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 *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
+ 19200000, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set DMIC cpu system clock\n");
+ return ret;
+ }
+ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set DMIC output clock\n");
+ return ret;
+ }
+ return 0;
+}
+
+static struct snd_soc_ops omap_abe_dmic_ops = {
+ .hw_params = omap_abe_dmic_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,
+ },
+};
+
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
+ /* Outputs */
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_SPK("Vibrator", NULL),
+
+ /* Inputs */
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
+ SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Routings for outputs */
+ {"Headset Stereophone", NULL, "HSOL"},
+ {"Headset Stereophone", NULL, "HSOR"},
+
+ {"Earphone Spk", NULL, "EP"},
+
+ {"Ext Spk", NULL, "HFL"},
+ {"Ext Spk", NULL, "HFR"},
+
+ {"Line Out", NULL, "AUXL"},
+ {"Line Out", NULL, "AUXR"},
+
+ {"Vibrator", NULL, "VIBRAL"},
+ {"Vibrator", NULL, "VIBRAR"},
+
+ /* Routings for inputs */
+ {"HSMIC", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "Headset Mic Bias"},
+
+ {"MAINMIC", NULL, "Main Handset Mic"},
+ {"Main Handset Mic", NULL, "Main Mic Bias"},
+
+ {"SUBMIC", NULL, "Sub Handset Mic"},
+ {"Sub Handset Mic", NULL, "Main Mic Bias"},
+
+ {"AFML", NULL, "Line In"},
+ {"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);
+ int hs_trim;
+ int ret = 0;
+
+ /* 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, "Vinrator");
+ 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");
+
+ /*
+ * Configure McPDM offset cancellation based on the HSOTRIM value from
+ * twl6040.
+ */
+ hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
+ omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
+ TWL6040_HSF_TRIM_RIGHT(hs_trim));
+
+ /* Headset jack detection only if it is supported */
+ if (pdata->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);
+ twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Digital Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route dmic_audio_map[] = {
+ {"DMic", NULL, "Digital Mic"},
+ {"Digital Mic", NULL, "Digital Mic1 Bias"},
+};
+
+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;
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+ ARRAY_SIZE(dmic_dapm_widgets));
+ if (ret)
+ return ret;
+
+ return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
+ ARRAY_SIZE(dmic_audio_map));
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link twl6040_dmic_dai[] = {
+ {
+ .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,
+ },
+ {
+ .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,
+ },
+};
+
+static struct snd_soc_dai_link twl6040_only_dai[] = {
+ {
+ .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,
+ },
+};
+
+/* Audio machine driver */
+static struct snd_soc_card omap_abe_card = {
+ .owner = THIS_MODULE,
+
+ .dapm_widgets = twl6040_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static __devinit int omap_abe_probe(struct platform_device *pdev)
+{
+ struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
+ struct snd_soc_card *card = &omap_abe_card;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ if (!pdata) {
+ dev_err(&pdev->dev, "Missing pdata\n");
+ return -ENODEV;
+ }
+
+ if (pdata->card_name) {
+ card->name = pdata->card_name;
+ } else {
+ dev_err(&pdev->dev, "Card name is not provided\n");
+ return -ENODEV;
+ }
+
+ if (!pdata->mclk_freq) {
+ dev_err(&pdev->dev, "MCLK frequency missing\n");
+ return -ENODEV;
+ }
+
+ if (pdata->has_dmic) {
+ card->dai_link = twl6040_dmic_dai;
+ card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
+ } else {
+ card->dai_link = twl6040_only_dai;
+ card->num_links = ARRAY_SIZE(twl6040_only_dai);
+ }
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+
+ return ret;
+}
+
+static int __devexit omap_abe_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver omap_abe_driver = {
+ .driver = {
+ .name = "omap-abe-twl6040",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = omap_abe_probe,
+ .remove = __devexit_p(omap_abe_remove),
+};
+
+module_platform_driver(omap_abe_driver);
+
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-abe-twl6040");
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 0855c1cfa7f..4dcb5a7e40e 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -113,12 +113,10 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
mutex_lock(&dmic->mutex);
- if (!dai->active) {
- snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
+ if (!dai->active)
dmic->active = 1;
- } else {
+ else
ret = -EBUSY;
- }
mutex_unlock(&dmic->mutex);
@@ -445,6 +443,7 @@ static struct snd_soc_dai_driver omap_dmic_dai = {
.channels_max = 6,
.rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
.formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
},
.ops = &omap_dmic_dai_ops,
};
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 017371913ec..1287b870f22 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -744,17 +744,17 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
omap_mcbsp3_set_st_ch1_volume),
};
-int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id)
+int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai)
{
if (!cpu_is_omap34xx())
return -ENODEV;
- switch (mcbsp_id) {
+ switch (dai->id) {
case 1: /* McBSP 2 */
- return snd_soc_add_controls(codec, omap_mcbsp2_st_controls,
+ return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls,
ARRAY_SIZE(omap_mcbsp2_st_controls));
case 2: /* McBSP 3 */
- return snd_soc_add_controls(codec, omap_mcbsp3_st_controls,
+ return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls,
ARRAY_SIZE(omap_mcbsp3_st_controls));
default:
break;
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 65cde9d3807..476fe2add70 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -59,6 +59,6 @@ enum omap_mcbsp_div {
#define NUM_LINKS 5
#endif
-int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id);
+int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai);
#endif
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 0e25df4fa9e..39705561131 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -419,12 +419,14 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = {
.channels_max = 5,
.rates = OMAP_MCPDM_RATES,
.formats = OMAP_MCPDM_FORMATS,
+ .sig_bits = 24,
},
.capture = {
.channels_min = 1,
.channels_max = 3,
.rates = OMAP_MCPDM_RATES,
.formats = OMAP_MCPDM_FORMATS,
+ .sig_bits = 24,
},
.ops = &omap_mcpdm_dai_ops,
};
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index fada6ef43ee..58936c730a8 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -59,9 +59,8 @@ static int rx51_spk_func;
static int rx51_dmic_func;
static int rx51_jack_func;
-static void rx51_ext_control(struct snd_soc_codec *codec)
+static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
int hp = 0, hs = 0, tvout = 0;
switch (rx51_jack_func) {
@@ -102,11 +101,11 @@ static int rx51_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_card *card = rtd->card;
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
- rx51_ext_control(codec);
+ rx51_ext_control(&card->dapm);
return 0;
}
@@ -138,13 +137,13 @@ static int rx51_get_spk(struct snd_kcontrol *kcontrol,
static int rx51_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_spk_func == ucontrol->value.integer.value[0])
return 0;
rx51_spk_func = ucontrol->value.integer.value[0];
- rx51_ext_control(codec);
+ rx51_ext_control(&card->dapm);
return 1;
}
@@ -184,13 +183,13 @@ static int rx51_get_input(struct snd_kcontrol *kcontrol,
static int rx51_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_dmic_func == ucontrol->value.integer.value[0])
return 0;
rx51_dmic_func = ucontrol->value.integer.value[0];
- rx51_ext_control(codec);
+ rx51_ext_control(&card->dapm);
return 1;
}
@@ -206,13 +205,13 @@ static int rx51_get_jack(struct snd_kcontrol *kcontrol,
static int rx51_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (rx51_jack_func == ucontrol->value.integer.value[0])
return 0;
rx51_jack_func = ucontrol->value.integer.value[0];
- rx51_ext_control(codec);
+ rx51_ext_control(&card->dapm);
return 1;
}
@@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "LINE1R");
/* Add RX-51 specific controls */
- err = snd_soc_add_controls(codec, aic34_rx51_controls,
+ err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls,
ARRAY_SIZE(aic34_rx51_controls));
if (err < 0)
return err;
@@ -314,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
return err;
snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
- err = omap_mcbsp_st_add_controls(codec, 1);
+ err = omap_mcbsp_st_add_controls(rtd->cpu_dai);
if (err < 0)
return err;
@@ -335,7 +334,7 @@ static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm)
{
int err;
- err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb,
+ err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb,
ARRAY_SIZE(aic34_rx51_controlsb));
if (err < 0)
return err;
diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c
deleted file mode 100644
index 175ba9a04ed..00000000000
--- a/sound/soc/omap/sdp4430.c
+++ /dev/null
@@ -1,279 +0,0 @@
-/*
- * sdp4430.c -- SoC audio for TI OMAP4430 SDP
- *
- * 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/mfd/twl6040.h>
-#include <linux/module.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <plat/hardware.h>
-#include <plat/mux.h>
-
-#include "omap-dmic.h"
-#include "omap-mcpdm.h"
-#include "omap-pcm.h"
-#include "../codecs/twl6040.h"
-
-static int sdp4430_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 clk_id, freq;
- int ret;
-
- clk_id = twl6040_get_clk_id(rtd->codec);
- if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
- freq = 38400000;
- else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
- freq = 32768;
- else
- return -EINVAL;
-
- /* set the codec mclk */
- ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
- SND_SOC_CLOCK_IN);
- if (ret) {
- printk(KERN_ERR "can't set codec system clock\n");
- return ret;
- }
- return ret;
-}
-
-static struct snd_soc_ops sdp4430_ops = {
- .hw_params = sdp4430_hw_params,
-};
-
-static int sdp4430_dmic_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 *cpu_dai = rtd->cpu_dai;
- int ret = 0;
-
- ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
- 19200000, SND_SOC_CLOCK_IN);
- if (ret < 0) {
- printk(KERN_ERR "can't set DMIC cpu system clock\n");
- return ret;
- }
- ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
- SND_SOC_CLOCK_OUT);
- if (ret < 0) {
- printk(KERN_ERR "can't set DMIC output clock\n");
- return ret;
- }
- return 0;
-}
-
-static struct snd_soc_ops sdp4430_dmic_ops = {
- .hw_params = sdp4430_dmic_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,
- },
-};
-
-/* SDP4430 machine DAPM */
-static const struct snd_soc_dapm_widget sdp4430_twl6040_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_SPK("Earphone Spk", NULL),
- SND_SOC_DAPM_INPUT("FM Stereo In"),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- /* External Mics: MAINMIC, SUBMIC with bias*/
- {"MAINMIC", NULL, "Main Mic Bias"},
- {"SUBMIC", NULL, "Main Mic Bias"},
- {"Main Mic Bias", 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"},
-
- /* Earphone speaker */
- {"Earphone Spk", NULL, "EP"},
-
- /* Aux/FM Stereo In: AFML, AFMR */
- {"AFML", NULL, "FM Stereo In"},
- {"AFMR", NULL, "FM Stereo In"},
-};
-
-static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- int ret, hs_trim;
-
- /*
- * Configure McPDM offset cancellation based on the HSOTRIM value from
- * twl6040.
- */
- hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
- omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
- TWL6040_HSF_TRIM_RIGHT(hs_trim));
-
- /* 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 (machine_is_omap_4430sdp())
- twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
- else
- snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET);
-
- return ret;
-}
-
-static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Digital Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route dmic_audio_map[] = {
- {"DMic", NULL, "Digital Mic1 Bias"},
- {"Digital Mic1 Bias", NULL, "Digital Mic"},
-};
-
-static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets,
- ARRAY_SIZE(sdp4430_dmic_dapm_widgets));
- if (ret)
- return ret;
-
- return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
- ARRAY_SIZE(dmic_audio_map));
-}
-
-/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp4430_dai[] = {
- {
- .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 = sdp4430_twl6040_init,
- .ops = &sdp4430_ops,
- },
- {
- .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 = sdp4430_dmic_init,
- .ops = &sdp4430_dmic_ops,
- },
-};
-
-/* Audio machine driver */
-static struct snd_soc_card snd_soc_sdp4430 = {
- .name = "SDP4430",
- .owner = THIS_MODULE,
- .dai_link = sdp4430_dai,
- .num_links = ARRAY_SIZE(sdp4430_dai),
-
- .dapm_widgets = sdp4430_twl6040_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *sdp4430_snd_device;
-
-static int __init sdp4430_soc_init(void)
-{
- int ret;
-
- if (!machine_is_omap_4430sdp())
- return -ENODEV;
- printk(KERN_INFO "SDP4430 SoC init\n");
-
- sdp4430_snd_device = platform_device_alloc("soc-audio", -1);
- if (!sdp4430_snd_device) {
- printk(KERN_ERR "Platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430);
-
- ret = platform_device_add(sdp4430_snd_device);
- if (ret)
- goto err;
-
- return 0;
-
-err:
- printk(KERN_ERR "Unable to add platform device\n");
- platform_device_put(sdp4430_snd_device);
- return ret;
-}
-module_init(sdp4430_soc_init);
-
-static void __exit sdp4430_soc_exit(void)
-{
- platform_device_unregister(sdp4430_snd_device);
-}
-module_exit(sdp4430_soc_exit);
-
-MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
-MODULE_DESCRIPTION("ALSA SoC SDP4430");
-MODULE_LICENSE("GPL");
-
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-21 20:58:19 +0000