diff options
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/aess.h | 53 | ||||
| -rw-r--r-- | include/sound/compress_driver.h | 8 | ||||
| -rw-r--r-- | include/sound/core.h | 12 | ||||
| -rw-r--r-- | include/sound/cs4271.h | 15 | ||||
| -rw-r--r-- | include/sound/da7213.h | 52 | ||||
| -rwxr-xr-x | include/sound/max98090.h | 29 | ||||
| -rw-r--r-- | include/sound/memalloc.h | 2 | ||||
| -rw-r--r-- | include/sound/saif.h | 16 | ||||
| -rw-r--r-- | include/sound/sh_fsi.h | 70 | ||||
| -rw-r--r-- | include/sound/simple_card.h | 12 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 8 | ||||
| -rw-r--r-- | include/sound/soc.h | 6 | ||||
| -rw-r--r-- | include/sound/tlv320aic3x.h | 10 | ||||
| -rw-r--r-- | include/sound/wm2000.h | 3 | ||||
| -rw-r--r-- | include/sound/wm2200.h | 22 |
15 files changed, 217 insertions, 101 deletions
diff --git a/include/sound/aess.h b/include/sound/aess.h new file mode 100644 index 000000000000..cee0d09fadbd --- /dev/null +++ b/include/sound/aess.h @@ -0,0 +1,53 @@ +/* + * AESS IP block reset + * + * Copyright (C) 2012 Texas Instruments, Inc. + * Paul Walmsley + * + * 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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; 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 __SOUND_AESS_H__ +#define __SOUND_AESS_H__ + +#include <linux/kernel.h> +#include <linux/io.h> + +/* + * AESS_AUTO_GATING_ENABLE_OFFSET: offset in bytes of the AESS IP + * block's AESS_AUTO_GATING_ENABLE__1 register from the IP block's + * base address + */ +#define AESS_AUTO_GATING_ENABLE_OFFSET 0x07c + +/* Register bitfields in the AESS_AUTO_GATING_ENABLE__1 register */ +#define AESS_AUTO_GATING_ENABLE_SHIFT 0 + +/** + * aess_enable_autogating - enable AESS internal autogating + * @oh: struct omap_hwmod * + * + * Enable internal autogating on the AESS. This allows the AESS to + * indicate that it is idle to the OMAP PRCM. Returns 0. + */ +static inline void aess_enable_autogating(void __iomem *base) +{ + u32 v; + + /* Set AESS_AUTO_GATING_ENABLE__1.ENABLE to allow idle entry */ + v = 1 << AESS_AUTO_GATING_ENABLE_SHIFT; + writel(v, base + AESS_AUTO_GATING_ENABLE_OFFSET); +} + +#endif /* __SOUND_AESS_H__ */ diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912abacdf3..ff6c74153fa1 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/sound/core.h b/include/sound/core.h index 93896ad1fcdd..7cede2d6aa86 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -394,8 +394,11 @@ void __snd_printk(unsigned int level, const char *file, int line, #else /* !CONFIG_SND_DEBUG */ -#define snd_printd(fmt, args...) do { } while (0) -#define _snd_printd(level, fmt, args...) do { } while (0) +__printf(1, 2) +static inline void snd_printd(const char *format, ...) {} +__printf(2, 3) +static inline void _snd_printd(int level, const char *format, ...) {} + #define snd_BUG() do { } while (0) static inline int __snd_bug_on(int cond) { @@ -416,7 +419,8 @@ static inline int __snd_bug_on(int cond) #define snd_printdd(format, args...) \ __snd_printk(2, __FILE__, __LINE__, format, ##args) #else -#define snd_printdd(format, args...) do { } while (0) +__printf(1, 2) +static inline void snd_printdd(const char *format, ...) {} #endif @@ -454,6 +458,7 @@ struct snd_pci_quirk { #define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \ {_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), \ .value = (val), .name = (xname)} +#define snd_pci_quirk_name(q) ((q)->name) #else #define SND_PCI_QUIRK(vend,dev,xname,val) \ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val)} @@ -461,6 +466,7 @@ struct snd_pci_quirk { {_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)} #define SND_PCI_QUIRK_VENDOR(vend, xname, val) \ {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)} +#define snd_pci_quirk_name(q) "" #endif const struct snd_pci_quirk * diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index dd8c48d14ed9..70f45355acaa 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -20,6 +20,21 @@ struct cs4271_platform_data { int gpio_nreset; /* GPIO driving Reset pin, if any */ 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/da7213.h b/include/sound/da7213.h new file mode 100644 index 000000000000..673f5c39cbf2 --- /dev/null +++ b/include/sound/da7213.h @@ -0,0 +1,52 @@ +/* + * da7213.h - DA7213 ASoC Codec Driver Platform Data + * + * Copyright (c) 2013 Dialog Semiconductor + * + * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.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. + */ + +#ifndef _DA7213_PDATA_H +#define _DA7213_PDATA_H + +enum da7213_micbias_voltage { + DA7213_MICBIAS_1_6V = 0, + DA7213_MICBIAS_2_2V = 1, + DA7213_MICBIAS_2_5V = 2, + DA7213_MICBIAS_3_0V = 3, +}; + +enum da7213_dmic_data_sel { + DA7213_DMIC_DATA_LRISE_RFALL = 0, + DA7213_DMIC_DATA_LFALL_RRISE = 1, +}; + +enum da7213_dmic_samplephase { + DA7213_DMIC_SAMPLE_ON_CLKEDGE = 0, + DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE = 1, +}; + +enum da7213_dmic_clk_rate { + DA7213_DMIC_CLK_3_0MHZ = 0, + DA7213_DMIC_CLK_1_5MHZ = 1, +}; + +struct da7213_platform_data { + /* Mic Bias voltage */ + enum da7213_micbias_voltage micbias1_lvl; + enum da7213_micbias_voltage micbias2_lvl; + + /* DMIC config */ + enum da7213_dmic_data_sel dmic_data_sel; + enum da7213_dmic_samplephase dmic_samplephase; + enum da7213_dmic_clk_rate dmic_clk_rate; + + /* MCLK squaring config */ + bool mclk_squaring; +}; + +#endif /* _DA7213_PDATA_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/memalloc.h b/include/sound/memalloc.h index 844af65af626..cf15b8213df7 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -37,7 +37,7 @@ struct snd_dma_device { #ifndef snd_dma_pci_data #define snd_dma_pci_data(pci) (&(pci)->dev) #define snd_dma_isa_data() NULL -#define snd_dma_continuous_data(x) ((struct device *)(unsigned long)(x)) +#define snd_dma_continuous_data(x) ((struct device *)(__force unsigned long)(x)) #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/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/simple_card.h b/include/sound/simple_card.h index 4b62b8dc6a4f..6c74527d4926 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -14,21 +14,21 @@ #include <sound/soc.h> -struct asoc_simple_dai_init_info { +struct asoc_simple_dai { + const char *name; unsigned int fmt; - unsigned int cpu_daifmt; - unsigned int codec_daifmt; unsigned int sysclk; }; struct asoc_simple_card_info { const char *name; const char *card; - const char *cpu_dai; const char *codec; const char *platform; - const char *codec_dai; - struct asoc_simple_dai_init_info *init; /* for snd_link.init */ + + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; /* used in simple-card.c */ struct snd_soc_dai_link snd_link; 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 bc56738cb109..a6a059ca3874 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,8 +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 { @@ -1171,6 +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> diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index ffd9bc793105..9407fd00363b 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -46,6 +46,13 @@ enum { AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 }; +enum aic3x_micbias_voltage { + AIC3X_MICBIAS_OFF = 0, + AIC3X_MICBIAS_2_0V = 1, + AIC3X_MICBIAS_2_5V = 2, + AIC3X_MICBIAS_AVDDV = 3, +}; + struct aic3x_setup_data { unsigned int gpio_func[2]; }; @@ -53,6 +60,9 @@ struct aic3x_setup_data { struct aic3x_pdata { int gpio_reset; /* < 0 if not used */ struct aic3x_setup_data *setup; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; #endif diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h index aa388ca9ec64..4de81f41c90f 100644 --- a/include/sound/wm2000.h +++ b/include/sound/wm2000.h @@ -15,9 +15,6 @@ struct wm2000_platform_data { /** Filename for system-specific image to download to device. */ const char *download_file; - /** Divide MCLK by 2 for system clock? */ - unsigned int mclkdiv2:1; - /** Disable speech clarity enhancement, for use when an * external algorithm is used. */ unsigned int speech_enh_disable:1; diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55be7ffa..bc7ab1a4b480 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 1, + WM2200_MBIAS_LVL_1V8 = 2, + WM2200_MBIAS_LVL_1V9 = 3, + WM2200_MBIAS_LVL_2V0 = 4, + WM2200_MBIAS_LVL_2V2 = 5, + WM2200_MBIAS_LVL_2V4 = 6, + WM2200_MBIAS_LVL_2V5 = 7, + WM2200_MBIAS_LVL_2V6 = 8, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif |