diff options
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/Kbuild | 0 | ||||
| -rw-r--r-- | include/sound/cs4271.h | 17 | ||||
| -rwxr-xr-x | include/sound/max98090.h | 29 | ||||
| -rw-r--r-- | include/sound/saif.h | 16 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 8 | ||||
| -rw-r--r-- | include/sound/soc.h | 16 |
6 files changed, 60 insertions, 26 deletions
diff --git a/include/sound/Kbuild b/include/sound/Kbuild deleted file mode 100644 index e69de29bb2d1..000000000000 --- a/include/sound/Kbuild +++ /dev/null diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 6d9e15ed1dcf..70f45355acaa 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -19,7 +19,22 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ - int amutec_eq_bmutec:1; /* flag to enable AMUTEC=BMUTEC */ + bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ + + /* + * The CS4271 requires its LRCLK and MCLK to be stable before its RESET + * line is de-asserted. That also means that clocks cannot be changed + * without putting the chip back into hardware reset, which also requires + * a complete re-initialization of all registers. + * + * One (undocumented) workaround is to assert and de-assert the PDN bit + * in the MODE2 register. This workaround can be enabled with the + * following flag. + * + * Note that this is not needed in case the clocks are stable + * throughout the entire runtime of the codec. + */ + bool enable_soft_reset; }; #endif /* __CS4271_H */ diff --git a/include/sound/max98090.h b/include/sound/max98090.h new file mode 100755 index 000000000000..95efb13f8478 --- /dev/null +++ b/include/sound/max98090.h @@ -0,0 +1,29 @@ +/* + * Platform data for MAX98090 + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SOUND_MAX98090_PDATA_H__ +#define __SOUND_MAX98090_PDATA_H__ + +/* codec platform data */ +struct max98090_pdata { + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + unsigned int digmic_3_mode:1; + unsigned int digmic_4_mode:1; +}; + +#endif diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e16edf4..000000000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * 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 __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3953cea0ecfb..3d84808952b9 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -45,7 +45,7 @@ struct snd_compr_stream; * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ -#define SND_SOC_DAIFMT_GATED (2 << 4) /* clock is gated */ +#define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* * DAI hardware signal inversions. @@ -53,7 +53,7 @@ struct snd_compr_stream; * Specifies whether the DAI can also support inverted clocks for the specified * format. */ -#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ @@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction); struct snd_soc_dai_ops { /* @@ -157,6 +158,7 @@ struct snd_soc_dai_ops { * Called by soc-core to minimise any pops. */ int (*digital_mute)(struct snd_soc_dai *dai, int mute); + int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); /* * ALSA PCM audio operations - all optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index 769e27c774a3..a6a059ca3874 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -58,8 +58,9 @@ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -88,8 +89,9 @@ .info = snd_soc_info_volsw_range, \ .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .min = xmin,\ - .max = xmax, .platform_max = xmax, .invert = xinvert} } + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -904,8 +906,8 @@ struct snd_soc_dai_link { struct snd_pcm_hw_params *params); /* machine stream operations */ - struct snd_soc_ops *ops; - struct snd_soc_compr_ops *compr_ops; + const struct snd_soc_ops *ops; + const struct snd_soc_compr_ops *compr_ops; }; struct snd_soc_codec_conf { @@ -1169,6 +1171,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +unsigned int snd_soc_of_parse_daifmt(struct device_node *np, + const char *prefix); #include <sound/soc-dai.h> |