Merge commit 'takashi/topic/asoc' into for-2.6.31
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 22b729f..a997c2c 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -45,24 +45,6 @@
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
/*
- * DAI Left/Right Clocks.
- *
- * Specifies whether the DAI can support different samples for similtanious
- * playback and capture. This usually requires a seperate physical frame
- * clock for playback and capture.
- */
-#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- *
- * Time Division Multiplexing. Allows PCM data to be multplexed with other
- * data on the DAI.
- */
-#define SND_SOC_DAIFMT_TDM (1 << 6)
-
-/*
* DAI hardware signal inversions.
*
* Specifies whether the DAI can also support inverted clocks for the specified
@@ -96,6 +78,9 @@
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index b0d4af1..932299b 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -53,13 +53,13 @@
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "AC97 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &ac97_dai_ops,
};
EXPORT_SYMBOL_GPL(ac97_dai);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index ddb3b08..d7440a9 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -137,13 +137,13 @@
.channels_min = 2,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .formats = SND_SOC_STD_AC97_FMTS, },
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .formats = SND_SOC_STD_AC97_FMTS, },
};
EXPORT_SYMBOL_GPL(ad1980_dai);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index efa1a80..1a00e4b 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -396,6 +396,31 @@
static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+/* Vibra */
+/* Vibra audio path selection */
+static const char *twl4030_vibra_texts[] =
+ {"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
+
+static const struct soc_enum twl4030_vibra_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
+ ARRAY_SIZE(twl4030_vibra_texts),
+ twl4030_vibra_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
+SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
+
+/* Vibra path selection: local vibrator (PWM) or audio driven */
+static const char *twl4030_vibrapath_texts[] =
+ {"Local vibrator", "Audio"};
+
+static const struct soc_enum twl4030_vibrapath_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
+ ARRAY_SIZE(twl4030_vibrapath_texts),
+ twl4030_vibrapath_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
+SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
+
/* Left analog microphone selection */
static const char *twl4030_analoglmic_texts[] =
{"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
@@ -468,6 +493,10 @@
static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
+/* Analog bypass for Voice */
+static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
+
/* Digital bypass gain, 0 mutes the bypass */
static const unsigned int twl4030_dapm_dbypass_tlv[] = {
TLV_DB_RANGE_HEAD(2),
@@ -487,6 +516,18 @@
TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
twl4030_dapm_dbypass_tlv);
+/*
+ * Voice Sidetone GAIN volume control:
+ * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB)
+ */
+static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1);
+
+/* Digital bypass voice: sidetone (VUL -> VDL)*/
+static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
+ SOC_DAPM_SINGLE_TLV("Volume",
+ TWL4030_REG_VSTPGA, 0, 0x29, 0,
+ twl4030_dapm_dbypassv_tlv);
+
static int micpath_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -585,7 +626,7 @@
struct soc_mixer_control *m =
(struct soc_mixer_control *)w->kcontrols->private_value;
struct twl4030_priv *twl4030 = w->codec->private_data;
- unsigned char reg;
+ unsigned char reg, misc;
reg = twl4030_read_reg_cache(w->codec, m->reg);
@@ -597,14 +638,34 @@
else
twl4030->bypass_state &=
~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+ } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) {
+ /* Analog voice bypass */
+ if (reg & (1 << m->shift))
+ twl4030->bypass_state |= (1 << 4);
+ else
+ twl4030->bypass_state &= ~(1 << 4);
+ } else if (m->reg == TWL4030_REG_VSTPGA) {
+ /* Voice digital bypass */
+ if (reg)
+ twl4030->bypass_state |= (1 << 5);
+ else
+ twl4030->bypass_state &= ~(1 << 5);
} else {
/* Digital bypass */
if (reg & (0x7 << m->shift))
- twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+ twl4030->bypass_state |= (1 << (m->shift ? 7 : 6));
else
- twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+ twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6));
}
+ /* Enable master analog loopback mode if any analog switch is enabled*/
+ misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1);
+ if (twl4030->bypass_state & 0x1F)
+ misc |= TWL4030_FMLOOP_EN;
+ else
+ misc &= ~TWL4030_FMLOOP_EN;
+ twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc);
+
if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
if (twl4030->bypass_state)
twl4030_codec_mute(w->codec, 0);
@@ -831,6 +892,26 @@
ARRAY_SIZE(twl4030_rampdelay_texts),
twl4030_rampdelay_texts);
+/* Vibra H-bridge direction mode */
+static const char *twl4030_vibradirmode_texts[] = {
+ "Vibra H-bridge direction", "Audio data MSB",
+};
+
+static const struct soc_enum twl4030_vibradirmode_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
+ ARRAY_SIZE(twl4030_vibradirmode_texts),
+ twl4030_vibradirmode_texts);
+
+/* Vibra H-bridge direction */
+static const char *twl4030_vibradir_texts[] = {
+ "Positive polarity", "Negative polarity",
+};
+
+static const struct soc_enum twl4030_vibradir_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
+ ARRAY_SIZE(twl4030_vibradir_texts),
+ twl4030_vibradir_texts);
+
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
/* Common playback gain controls */
SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
@@ -897,6 +978,9 @@
0, 3, 5, 0, input_gain_tlv),
SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+
+ SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
+ SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
};
static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
@@ -924,6 +1008,7 @@
SND_SOC_DAPM_OUTPUT("CARKITR"),
SND_SOC_DAPM_OUTPUT("HFL"),
SND_SOC_DAPM_OUTPUT("HFR"),
+ SND_SOC_DAPM_OUTPUT("VIBRA"),
/* DACs */
SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
@@ -935,7 +1020,7 @@
SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback",
- TWL4030_REG_AVDAC_CTL, 4, 0),
+ SND_SOC_NOPM, 0, 0),
/* Analog PGAs */
SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
@@ -962,6 +1047,9 @@
SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassl2_control,
bypass_event, SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_abypassv_control,
+ bypass_event, SND_SOC_DAPM_POST_REG),
/* Digital bypasses */
SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
@@ -970,6 +1058,9 @@
SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_dbypassr_control, bypass_event,
SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_dbypassv_control, bypass_event,
+ SND_SOC_DAPM_POST_REG),
SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
0, 0, NULL, 0),
@@ -979,6 +1070,8 @@
2, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
3, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_AVDAC_CTL,
+ 4, 0, NULL, 0),
/* Output MIXER controls */
/* Earpiece */
@@ -1016,6 +1109,11 @@
SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
&twl4030_dapm_handsfreer_control, handsfree_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ /* Vibra */
+ SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
+ &twl4030_dapm_vibra_control),
+ SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_vibrapath_control),
/* Introducing four virtual ADC, since TWL4030 have four channel for
capture */
@@ -1067,13 +1165,13 @@
{"Analog R1 Playback Mixer", NULL, "DAC Right1"},
{"Analog L2 Playback Mixer", NULL, "DAC Left2"},
{"Analog R2 Playback Mixer", NULL, "DAC Right2"},
+ {"Analog Voice Playback Mixer", NULL, "DAC Voice"},
{"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
{"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
{"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
{"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
-
- {"VDL_APGA", NULL, "DAC Voice"},
+ {"VDL_APGA", NULL, "Analog Voice Playback Mixer"},
/* Internal playback routings */
/* Earpiece */
@@ -1117,6 +1215,11 @@
{"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"},
{"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"},
{"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"},
+ /* Vibra */
+ {"Vibra Mux", "AudioL1", "DAC Left1"},
+ {"Vibra Mux", "AudioR1", "DAC Right1"},
+ {"Vibra Mux", "AudioL2", "DAC Left2"},
+ {"Vibra Mux", "AudioR2", "DAC Right2"},
/* outputs */
{"OUTL", NULL, "ARXL2_APGA"},
@@ -1130,6 +1233,8 @@
{"CARKITR", NULL, "CarkitR Mixer"},
{"HFL", NULL, "HandsfreeL Mux"},
{"HFR", NULL, "HandsfreeR Mux"},
+ {"Vibra Route", "Audio", "Vibra Mux"},
+ {"VIBRA", NULL, "Vibra Route"},
/* Capture path */
{"Analog Left Capture Route", "Main mic", "MAINMIC"},
@@ -1169,18 +1274,22 @@
{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+ {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"},
{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+ {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"},
/* Digital bypass routes */
{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+ {"Voice Digital Loopback", "Volume", "TX2 Capture Route"},
{"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
{"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+ {"Analog Voice Playback Mixer", NULL, "Voice Digital Loopback"},
};
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index c2d1a7a..fa88b46 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -282,14 +282,14 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SND_SOC_STD_AC97_FMTS,
},
.ops = &wm9705_dai_ops,
},
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 765cf1e..550c903 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -534,13 +534,13 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_hifi,
},
{
@@ -550,7 +550,7 @@
.channels_min = 1,
.channels_max = 1,
.rates = WM9712_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_aux,
}
};
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index a6feb784..d1744e9 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1040,13 +1040,13 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_hifi,
},
{
@@ -1056,7 +1056,7 @@
.channels_min = 1,
.channels_max = 1,
.rates = WM9713_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_aux,
},
{
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ad8a10f..eb75a1c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -134,3 +134,12 @@
help
Say Y if you want to add support for SoC audio on the
MIO A701.
+
+config SND_PXA2XX_SOC_IMOTE2
+ tristate "SoC Audio support for IMote 2"
+ depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8940
+ help
+ Say Y if you want to add support for SoC audio on the
+ IMote 2.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 4b90c3c..6e096b4 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -22,6 +22,7 @@
snd-soc-zylonite-objs := zylonite.o
snd-soc-magician-objs := magician.o
snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-imote2-objs := imote2.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -35,3 +36,4 @@
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
+obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
new file mode 100644
index 0000000..405587a
--- /dev/null
+++ b/sound/soc/pxa/imote2.c
@@ -0,0 +1,114 @@
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm8940.h"
+#include "pxa2xx-i2s.h"
+#include "pxa2xx-pcm.h"
+
+static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* CPU should be clock master */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk,
+ SND_SOC_CLOCK_OUT);
+
+ return ret;
+}
+
+static struct snd_soc_ops imote2_asoc_ops = {
+ .hw_params = imote2_asoc_hw_params,
+};
+
+static struct snd_soc_dai_link imote2_dai = {
+ .name = "WM8940",
+ .stream_name = "WM8940",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8940_dai,
+ .ops = &imote2_asoc_ops,
+};
+
+static struct snd_soc_card snd_soc_imote2 = {
+ .name = "Imote2",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = &imote2_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device imote2_snd_devdata = {
+ .card = &snd_soc_imote2,
+ .codec_dev = &soc_codec_dev_wm8940,
+};
+
+static struct platform_device *imote2_snd_device;
+
+static int __init imote2_asoc_init(void)
+{
+ int ret;
+
+ if (!machine_is_intelmote2())
+ return -ENODEV;
+ imote2_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!imote2_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata);
+ imote2_snd_devdata.dev = &imote2_snd_device->dev;
+ ret = platform_device_add(imote2_snd_device);
+ if (ret)
+ platform_device_put(imote2_snd_device);
+
+ return ret;
+}
+module_init(imote2_asoc_init);
+
+static void __exit imote2_asoc_exit(void)
+{
+ platform_device_unregister(imote2_snd_device);
+}
+module_exit(imote2_asoc_exit);
+
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_DESCRIPTION("ALSA SoC Imote 2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index ab680aa..972c276 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -37,6 +37,20 @@
#include "s3c-i2s-v2.h"
+#undef S3C_IIS_V2_SUPPORTED
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifndef S3C_IIS_V2_SUPPORTED
+#error Unsupported CPU model
+#endif
+
#define S3C2412_I2S_DEBUG_CON 0
static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
@@ -75,7 +89,7 @@
/* Turn on or off the transmission path. */
-void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
{
void __iomem *regs = i2s->regs;
u32 fic, con, mod;
@@ -105,7 +119,9 @@
break;
default:
- dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
+ break;
}
writel(con, regs + S3C2412_IISCON);
@@ -132,7 +148,9 @@
break;
default:
- dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
+ break;
}
writel(mod, regs + S3C2412_IISMOD);
@@ -143,9 +161,8 @@
dbg_showcon(__func__, con);
pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
}
-EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
-void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
{
void __iomem *regs = i2s->regs;
u32 fic, con, mod;
@@ -175,7 +192,8 @@
break;
default:
- dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
}
writel(mod, regs + S3C2412_IISMOD);
@@ -199,7 +217,8 @@
break;
default:
- dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
}
writel(con, regs + S3C2412_IISCON);
@@ -209,7 +228,6 @@
fic = readl(regs + S3C2412_IISFIC);
pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
}
-EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
/*
* Wait for the LR signal to allow synchronisation to the L/R clock
@@ -266,7 +284,7 @@
*/
#define IISMOD_MASTER_MASK (1 << 11)
#define IISMOD_SLAVE (1 << 11)
-#define IISMOD_MASTER (0x0)
+#define IISMOD_MASTER (0 << 11)
#endif
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -281,7 +299,7 @@
iismod |= IISMOD_MASTER;
break;
default:
- pr_debug("unknwon master/slave format\n");
+ pr_err("unknwon master/slave format\n");
return -EINVAL;
}
@@ -298,7 +316,7 @@
iismod |= S3C2412_IISMOD_SDF_IIS;
break;
default:
- pr_debug("Unknown data format\n");
+ pr_err("Unknown data format\n");
return -EINVAL;
}
@@ -327,6 +345,7 @@
iismod = readl(i2s->regs + S3C2412_IISMOD);
pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
iismod |= S3C2412_IISMOD_8BIT;
@@ -335,6 +354,25 @@
iismod &= ~S3C2412_IISMOD_8BIT;
break;
}
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+ iismod &= ~0x606;
+ /* Sample size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ /* 8 bit sample, 16fs BCLK */
+ iismod |= 0x2004;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ /* 16 bit sample, 32fs BCLK */
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ /* 24 bit sample, 48fs BCLK */
+ iismod |= 0x4002;
+ break;
+ }
+#endif
writel(iismod, i2s->regs + S3C2412_IISMOD);
pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
@@ -489,6 +527,8 @@
unsigned int best_rate = 0;
unsigned int best_deviation = INT_MAX;
+ pr_debug("Input clock rate %ldHz\n", clkrate);
+
if (fstab == NULL)
fstab = iis_fs_tab;
@@ -539,12 +579,31 @@
unsigned long base)
{
struct device *dev = &pdev->dev;
+ unsigned int iismod;
i2s->dev = dev;
/* record our i2s structure for later use in the callbacks */
dai->private_data = i2s;
+ if (!base) {
+ struct resource *res = platform_get_resource(pdev,
+ IORESOURCE_MEM,
+ 0);
+ if (!res) {
+ dev_err(dev, "Unable to get register resource\n");
+ return -ENXIO;
+ }
+
+ if (!request_mem_region(res->start, resource_size(res),
+ "s3c64xx-i2s-v4")) {
+ dev_err(dev, "Unable to request register region\n");
+ return -EBUSY;
+ }
+
+ base = res->start;
+ }
+
i2s->regs = ioremap(base, 0x100);
if (i2s->regs == NULL) {
dev_err(dev, "cannot ioremap registers\n");
@@ -560,12 +619,16 @@
clk_enable(i2s->iis_pclk);
+ /* Mark ourselves as in TXRX mode so we can run through our cleanup
+ * process without warnings. */
+ iismod = readl(i2s->regs + S3C2412_IISMOD);
+ iismod |= S3C2412_IISMOD_MODE_TXRX;
+ writel(iismod, i2s->regs + S3C2412_IISMOD);
s3c2412_snd_txctrl(i2s, 0);
s3c2412_snd_rxctrl(i2s, 0);
return 0;
}
-
EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
#ifdef CONFIG_PM
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 1345fbd..3c06c40 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -108,14 +108,13 @@
return 0;
}
-
-unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
{
struct s3c_i2sv2_info *i2s = to_info(dai);
- return clk_get_rate(i2s->iis_cclk);
+ return i2s->iis_cclk;
}
-EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
static int s3c64xx_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
@@ -147,7 +146,8 @@
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define S3C64XX_I2S_FMTS \
- (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
.set_sysclk = s3c64xx_i2s_set_sysclk,
@@ -215,13 +215,12 @@
i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus");
if (IS_ERR(i2s->iis_cclk)) {
- dev_err(&pdev->dev, "failed to get audio-bus");
+ dev_err(&pdev->dev, "failed to get audio-bus\n");
ret = PTR_ERR(i2s->iis_cclk);
goto err;
}
- ret = s3c_i2sv2_probe(pdev, dai, i2s,
- dai->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
+ ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
if (ret)
goto err_clk;
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index 597822a..02148ce 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -15,6 +15,8 @@
#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H
#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
+struct clk;
+
#include "s3c-i2s-v2.h"
#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK
@@ -26,6 +28,6 @@
extern struct snd_soc_dai s3c64xx_i2s_dai[];
-extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai);
#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */