diff options
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/soc-dai.h | 14 | ||||
| -rw-r--r-- | include/sound/soc-dapm.h | 17 | ||||
| -rw-r--r-- | include/sound/soc.h | 15 | ||||
| -rw-r--r-- | include/sound/tlv320dac33-plat.h | 20 | ||||
| -rw-r--r-- | include/sound/tpa6130a2-plat.h | 30 |
5 files changed, 87 insertions, 9 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..ca24e7f 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 c1410e3..c5c95e1d 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 475cb7e..0d7718f9 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 0000000..5858d06 --- /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 0000000..e8c901e --- /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 |
