diff options
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/cs4271.h | 17 | ||||
| -rwxr-xr-x | include/sound/max98090.h | 29 | ||||
| -rw-r--r-- | include/sound/sh_fsi.h | 70 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 6 | ||||
| -rw-r--r-- | include/sound/soc.h | 14 |
5 files changed, 61 insertions, 75 deletions
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/sh_fsi.h b/include/sound/sh_fsi.h index cc1c919c6436..7a9710b4b799 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -11,82 +11,20 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ - -#define FSI_PORT_A 0 -#define FSI_PORT_B 1 - #include <linux/clk.h> #include <sound/soc.h> /* - * flags format - * - * 0x00000CBA - * - * A: inversion - * B: format mode - * C: chip specific - * D: clock selecter if master mode + * flags */ - -/* A: clock inversion */ -#define SH_FSI_INVERSION_MASK 0x0000000F -#define SH_FSI_LRM_INV (1 << 0) -#define SH_FSI_BRM_INV (1 << 1) -#define SH_FSI_LRS_INV (1 << 2) -#define SH_FSI_BRS_INV (1 << 3) - -/* B: format mode */ -#define SH_FSI_FMT_MASK 0x000000F0 -#define SH_FSI_FMT_DAI (0 << 4) -#define SH_FSI_FMT_SPDIF (1 << 4) - -/* C: chip specific */ -#define SH_FSI_OPTION_MASK 0x00000F00 -#define SH_FSI_ENABLE_STREAM_MODE (1 << 8) /* for 16bit data */ - -/* D: clock selecter if master mode */ -#define SH_FSI_CLK_MASK 0x0000F000 -#define SH_FSI_CLK_EXTERNAL (0 << 12) -#define SH_FSI_CLK_CPG (1 << 12) /* FSIxCK + FSI-DIV */ - -/* - * set_rate return value - * - * see ACKMD/BPFMD on - * ACK_MD (FSI2) - * CKG1 (FSI) - * - * err : return value < 0 - * no change : return value == 0 - * change xMD : return value > 0 - * - * 0x-00000AB - * - * A: ACKMD value - * B: BPFMD value - */ - -#define SH_FSI_ACKMD_MASK (0xF << 0) -#define SH_FSI_ACKMD_512 (1 << 0) -#define SH_FSI_ACKMD_256 (2 << 0) -#define SH_FSI_ACKMD_128 (3 << 0) -#define SH_FSI_ACKMD_64 (4 << 0) -#define SH_FSI_ACKMD_32 (5 << 0) - -#define SH_FSI_BPFMD_MASK (0xF << 4) -#define SH_FSI_BPFMD_512 (1 << 4) -#define SH_FSI_BPFMD_256 (2 << 4) -#define SH_FSI_BPFMD_128 (3 << 4) -#define SH_FSI_BPFMD_64 (4 << 4) -#define SH_FSI_BPFMD_32 (5 << 4) -#define SH_FSI_BPFMD_16 (6 << 4) +#define SH_FSI_FMT_SPDIF (1 << 0) /* spdif for HDMI */ +#define SH_FSI_ENABLE_STREAM_MODE (1 << 1) /* for 16bit data */ +#define SH_FSI_CLK_CPG (1 << 2) /* FSIxCK + FSI-DIV */ struct sh_fsi_port_info { unsigned long flags; int tx_id; int rx_id; - int (*set_rate)(struct device *dev, int rate, int enable); }; struct sh_fsi_platform_info { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3953cea0ecfb..e30a00c14196 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -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..c76d0242c2a8 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 { |
