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-rw-r--r--include/sound/soc-dai.h14
-rw-r--r--include/sound/soc-dapm.h17
-rw-r--r--include/sound/soc.h15
-rw-r--r--include/sound/tlv320dac33-plat.h20
-rw-r--r--include/sound/tpa6130a2-plat.h30
5 files changed, 87 insertions, 9 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 97ca9af414d..ca24e7f7a3f 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -30,6 +30,7 @@ struct snd_pcm_substream;
#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
+#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
@@ -106,7 +107,7 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div);
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
/* Digital Audio interface formatting */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
@@ -114,6 +115,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
+
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
@@ -136,8 +141,8 @@ struct snd_soc_dai_ops {
*/
int (*set_sysclk)(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir);
- int (*set_pll)(struct snd_soc_dai *dai,
- int pll_id, unsigned int freq_in, unsigned int freq_out);
+ int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/*
@@ -148,6 +153,9 @@ struct snd_soc_dai_ops {
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width);
+ int (*set_channel_map)(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c1410e3191e..c5c95e1da65 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -206,6 +206,12 @@
.get = snd_soc_dapm_get_enum_double, \
.put = snd_soc_dapm_put_enum_double, \
.private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_VIRT(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_dapm_get_enum_virt, \
+ .put = snd_soc_dapm_put_enum_virt, \
+ .private_value = (unsigned long)&xenum }
#define SOC_DAPM_VALUE_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
@@ -260,6 +266,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
@@ -333,6 +343,10 @@ struct snd_soc_dapm_route {
const char *sink;
const char *control;
const char *source;
+
+ /* Note: currently only supported for links where source is a supply */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
};
/* dapm audio path between two widgets */
@@ -349,6 +363,9 @@ struct snd_soc_dapm_path {
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
+ int (*connected)(struct snd_soc_dapm_widget *source,
+ struct snd_soc_dapm_widget *sink);
+
struct list_head list_source;
struct list_head list_sink;
struct list_head list;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 475cb7ed6be..0d7718f9280 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -223,15 +223,15 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
int addr_bits, int data_bits,
enum snd_soc_control_type control);
-#ifdef CONFIG_PM
-int snd_soc_suspend_device(struct device *dev);
-int snd_soc_resume_device(struct device *dev);
-#endif
-
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_init_card(struct snd_soc_device *socdev);
+
+/* Utility functions to get clock rates from various things */
+int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
+int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots);
+int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
/* set runtime hw params */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@@ -333,6 +333,8 @@ struct snd_soc_jack_gpio {
int debounce_time;
struct snd_soc_jack *jack;
struct work_struct work;
+
+ int (*jack_status_check)(void);
};
#endif
@@ -413,6 +415,7 @@ struct snd_soc_codec {
unsigned int num_dai;
#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_codec_root;
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
struct dentry *debugfs_dapm;
diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
new file mode 100644
index 00000000000..5858d06a7ff
--- /dev/null
+++ b/include/sound/tlv320dac33-plat.h
@@ -0,0 +1,20 @@
+/*
+ * Platform header for Texas Instruments TLV320DAC33 codec driver
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *
+ * Copyright: (C) 2009 Nokia Corporation
+ *
+ * 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.
+ */
+
+#ifndef __TLV320DAC33_PLAT_H
+#define __TLV320DAC33_PLAT_H
+
+struct tlv320dac33_platform_data {
+ int power_gpio;
+};
+
+#endif /* __TLV320DAC33_PLAT_H */
diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
new file mode 100644
index 00000000000..e8c901e749d
--- /dev/null
+++ b/include/sound/tpa6130a2-plat.h
@@ -0,0 +1,30 @@
+/*
+ * TPA6130A2 driver platform header
+ *
+ * Copyright (C) Nokia Corporation
+ *
+ * Written by Peter Ujfalusi <peter.ujfalusi@nokia.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 TPA6130A2_PLAT_H
+#define TPA6130A2_PLAT_H
+
+struct tpa6130a2_platform_data {
+ int power_gpio;
+};
+
+#endif
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-24 21:34:17 +0000