From d61100bbd18e8b3fc9406be55354dabd5e7525ec Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 14 Dec 2012 15:16:58 +0900 Subject: ASoC: wm2000: Use clock API integration to configure MCLK divisor Since we are now using the clock API integration to manage MCLK we can now use clk_get_rate() to determine if we need to divide MCLK without relying on platform data. Signed-off-by: Mark Brown --- include/sound/wm2000.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'include') diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h index aa388ca9ec6..4de81f41c90 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; -- cgit v1.2.3-18-g5258 From fd23fb9f6bfd43a6e62b2646d18d5ca3edc3ebe3 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 10 Dec 2012 10:30:04 +0100 Subject: ALSA: ASoC: cs4271: add optional soft reset workaround 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 patch adds a new flag to both the DT bindings as well as to the platform data to enable that workaround. Signed-off-by: Daniel Mack Acked-by: Alexander Sverdlin Signed-off-by: Mark Brown --- include/sound/cs4271.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'include') diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index dd8c48d14ed..70f45355aca 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 */ -- cgit v1.2.3-18-g5258 From 6cbdbffba19620db77de38094f407b6f21d3f10c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 16 Dec 2012 22:12:21 -0800 Subject: ASoC: fsi: remove platform depended .set_rate() callback support ab6f6d85210c4d0265cf48e9958c04e08595055a (ASoC: fsi: add master clock control functions) added driver level clock control functions. And now, platform depended .set_rate() is no longer needed. This patch removed unnecessary .set_rate() platform callback support. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/sh_fsi.h | 33 --------------------------------- 1 file changed, 33 deletions(-) (limited to 'include') diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index cc1c919c643..66285e1e340 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -50,43 +50,10 @@ #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) - 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 { -- cgit v1.2.3-18-g5258 From abca75814a82c0c53c0a8ec7fa1300c133bc4f01 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 16 Dec 2012 22:12:42 -0800 Subject: ASoC: fsi: remove SH_FSI_xxx_INV flags 3449f5fab8c51e37a8a48bc2516588c615373191 (ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support) added clock inversion support via snd_soc_dai_set_fmt(). Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info() from fsi driver, and modified platform settings to use new style. Then, it cleaned up meaningless settings from platform. Signed-off-by: Kuninori Morimoto Acked-by: Simon Horman Signed-off-by: Mark Brown --- include/sound/sh_fsi.h | 7 ------- 1 file changed, 7 deletions(-) (limited to 'include') diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 66285e1e340..43ac2858192 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -29,13 +29,6 @@ * D: clock selecter if master mode */ -/* 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) -- cgit v1.2.3-18-g5258 From 5d0bfc5eb9f57b319d7cd6a1d5543c8287c77812 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 16 Dec 2012 22:12:55 -0800 Subject: ASoC: fsi: cleanup sh_fsi.h FSI driver's flag usage was changed/removed by 3449f5fab8c51e37a8a48bc2516588c615373191 (ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support) ab6f6d85210c4d0265cf48e9958c04e08595055a (ASoC: fsi: add master clock control functions) And unused flags had been removed on FSI driver, but the definition had been kept to avoid compile error. It is possible to cleanup sh_fsi.h now. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/sh_fsi.h | 30 ++++-------------------------- 1 file changed, 4 insertions(+), 26 deletions(-) (limited to 'include') diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 43ac2858192..7a9710b4b79 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -11,37 +11,15 @@ * 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 #include /* - * flags format - * - * 0x00000CBA - * - * A: inversion - * B: format mode - * C: chip specific - * D: clock selecter if master mode + * flags */ - -/* 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 */ +#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; -- cgit v1.2.3-18-g5258 From 4498a3cae5012979bbf3be2064c5ca00fe29109b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 14 Nov 2012 18:28:58 -0200 Subject: ASoC: mxs-saif: Remove platform data All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam Acked-by: Dong Aisheng Acked-by: Shawn Guo Signed-off-by: Mark Brown --- include/sound/saif.h | 16 ---------------- 1 file changed, 16 deletions(-) delete mode 100644 include/sound/saif.h (limited to 'include') diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e16edf..00000000000 --- 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 -- cgit v1.2.3-18-g5258 From a4a2992c531f6ca0aa00ce0deb31e51c1b7ae69b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Jan 2013 16:49:11 -0800 Subject: ASoC: simple-card: add asoc_simple_dai for initializing Current simple-card driver calls asoc_simple_card_dai_init() if platform had a asoc_simple_card_dai_init pointer. And, this initialization function works only when platform has an applicable initial value for each dai settings. And basically, almost all sound card requires certain initialization. This means that almost all platform has initialization settings, and driver do nothing if it doesn't have settings. And additionally, current simple-card supports sysclk settings but it was only for codec. In order to abolish deviation between cpu and codec, and in order to simplify processing, this patch adds asoc_simple_dai, and removed pointless struct asoc_simple_dai_init_info which was trigger of calling asoc_simple_card_dai_init(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card.h | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'include') diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 4b62b8dc6a4..6c74527d492 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -14,21 +14,21 @@ #include -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; -- cgit v1.2.3-18-g5258 From 13aec722f3c14aa6019c800465aa3ddd3638d305 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 10 Jan 2013 17:06:15 +0100 Subject: ASoC: Constify ops and compr_ops fields of snd_soc_dai_link The core does not modify these fields, so they can be made const. This allows drivers to declare their op tables as const. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 769e27c774a..bedf3dabdbd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -904,8 +904,8 @@ 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 { -- cgit v1.2.3-18-g5258 From 5d163336a77af9c1b4d6d08cbc8b1279df5f579e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 15 Jan 2013 20:18:23 -0800 Subject: ASoC: SND_SOC_DAIFMT_NB_NF become 0 as default settings Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 << 8), but normal bit clock / normal frame should be default settings (= 0). This patch fixup SND_SOC_DAIFMT_NB_NF as (0 << 8). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3953cea0ecf..90dc00434da 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 */ -- cgit v1.2.3-18-g5258 From c94aa30edac4d328674e9c127918317009d30c1a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Jan 2013 16:35:14 +0900 Subject: ASoC: arizona: Allow number of channels clocked to be restricted Place a cap on the number of channels clocks are generated for. This is intended for use with systems which have the WM5102 master an I2S bus with multiple data lines. Signed-off-by: Mark Brown --- include/linux/mfd/arizona/pdata.h | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'include') diff --git a/include/linux/mfd/arizona/pdata.h b/include/linux/mfd/arizona/pdata.h index 8b1d1daaae1..ec3e2a2a6d7 100644 --- a/include/linux/mfd/arizona/pdata.h +++ b/include/linux/mfd/arizona/pdata.h @@ -62,6 +62,8 @@ #define ARIZONA_MAX_OUTPUT 6 +#define ARIZONA_MAX_AIF 3 + #define ARIZONA_HAP_ACT_ERM 0 #define ARIZONA_HAP_ACT_LRA 2 @@ -96,6 +98,13 @@ struct arizona_pdata { /** Pin state for GPIO pins */ int gpio_defaults[ARIZONA_MAX_GPIO]; + /** + * Maximum number of channels clocks will be generated for, + * useful for systems where and I2S bus with multiple data + * lines is mastered. + */ + int max_channels_clocked[ARIZONA_MAX_AIF]; + /** GPIO for mic detection polarity */ int micd_pol_gpio; -- cgit v1.2.3-18-g5258 From 86b2723725a2e186f5699d97cb20014fa893931f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Jan 2013 10:54:07 +0100 Subject: ALSA: Make snd_printd() and snd_printdd() inline MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Because currently snd_printd() and snd_printdd() macros are expanded to empty when CONFIG_SND_DEBUG=n, a compile warning like below appears sometimes, and we had to covert it by ugly ifdefs: sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’: sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable] For "fixing" these issues better, this patch replaces snd_printd() and snd_printdd() definitions with empty inline functions instead of macros. This should have the same effect but shut up warnings like above. But since we had already put ifdefs, changing to inline functions would trigger compile errors. So, such ifdefs is removed in this patch. In addition, snd_pci_quirk name field is defined only when CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in snd_printdd() argument triggers the build errors, too. For avoiding these errors, introduce a new macro snd_pci_quirk_name() that is defined no matter how the debug option is set. Reported-by: Stratos Karafotis Signed-off-by: Takashi Iwai --- include/sound/core.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index 93896ad1fcd..7cede2d6aa8 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 * -- cgit v1.2.3-18-g5258 From a7930ed458afeacb029cee2b22f77b2a15472ad6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 14 Jan 2013 18:36:04 -0800 Subject: ASoC: add snd_soc_of_parse_daifmt() for DeviceTree This patch adds snd_soc_of_parse_daifmt() and supports below style on DT. [prefix]format = "i2c"; [prefix]clock-gating = "continuous"; [prefix]bitclock-inversion; [prefix]bitclock-master; [prefix]frame-master; Each driver can use specific [prefix] (ex simple-card,cpu,dai,format = xxx;) This sample will be SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index bc56738cb10..571562340e5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -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 -- cgit v1.2.3-18-g5258 From 019d80db5727707faa2108fcd4fbbfac9defb3a6 Mon Sep 17 00:00:00 2001 From: Antonio Ospite Date: Tue, 29 Jan 2013 12:56:26 +0100 Subject: ALSA: Force a cast to silence a warning from "sparse" Some audio drivers are calling snd_dma_continuous_data(GFP_KERNEL) which makes "sparse" give a warning: $ make C=2 M=sound/usb modules ... sound/usb/6fire/pcm.c:625:25: warning: cast from restricted gfp_t sound/usb/caiaq/audio.c:845:41: warning: cast from restricted gfp_t sound/usb/usx2y/usbusx2yaudio.c:997:54: warning: cast from restricted gfp_t sound/usb/usx2y/usbusx2yaudio.c:1001:54: warning: cast from restricted gfp_t sound/usb/usx2y/usx2yhwdeppcm.c:774:54: warning: cast from restricted gfp_t sound/usb/usx2y/usx2yhwdeppcm.c:778:54: warning: cast from restricted gfp_t Add __force to the cast to silence the warning. Signed-off-by: Antonio Ospite Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 844af65af62..cf15b8213df 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 -- cgit v1.2.3-18-g5258 From eef28e10821fb671ba797a41e7cf44e3d244e32e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 29 Jan 2013 21:03:13 -0800 Subject: ASoC: SND_SOC_DAIFMT_GATED become 0 as default settings Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 << 4), but gated clock should be default settings (= 0). This patch fixup SND_SOC_DAIFMT_GATED as (0 << 4). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3953cea0ecf..4dbd3e78eb8 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. -- cgit v1.2.3-18-g5258 From e2e8bfdf61573c98162d1112b971d8d00f00fcf8 Mon Sep 17 00:00:00 2001 From: Hebbar Gururaja Date: Thu, 31 Jan 2013 18:23:04 +0530 Subject: ASoC: tlv320aic3x: Convert mic bias to a supply widget Convert MicBias widgets to supply widget. On tlv320aic3x, Mic bias power on/off shares the same register bits with output mic bias voltage. So, when power on mic bias, we need reclaim it to voltage value. Provide a new platform data so that the micbias voltage can be sent according to board requirement. Now since tlv320aic3x codec driver is DT aware, update dt files and functions to handle this new "micbias-vg" platform data. Because of sharing of bits, when enabling the micbias, voltage also needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD macro to create an event to handle this. Since micbias is converted to supply widget, updated machine drivers as well. This change is runtime tested on da850-evm with audio loopback (arecord|aplay) for confirmation. Signed-off-by: Hebbar Gururaja Signed-off-by: Mark Brown --- include/sound/tlv320aic3x.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include') diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index ffd9bc79310..9407fd00363 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 -- cgit v1.2.3-18-g5258 From 1a786243235b8a8f4762ee57f185dadd97794fa4 Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Tue, 5 Feb 2013 14:40:44 +0000 Subject: ASoC: wm2200: Provide platform data for MICBIAS configuration Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- include/sound/wm2200.h | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55be7ff..bc7ab1a4b48 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 -- cgit v1.2.3-18-g5258 From 685e42154dcf3f6c0a52c115bd15e3d28ad8621b Mon Sep 17 00:00:00 2001 From: Jerry Wong Date: Wed, 6 Feb 2013 11:06:37 -0800 Subject: ASoC: Replace max98090 Device Driver This patch completes the replacement of the existing max98090 driver, by installing a more complete driver. Signed-off-by: Jerry Wong Tested-by: Matthew Mowdy Reviewed-by: Ralph Birt Signed-off-by: Mark Brown --- include/sound/max98090.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100755 include/sound/max98090.h (limited to 'include') diff --git a/include/sound/max98090.h b/include/sound/max98090.h new file mode 100755 index 00000000000..95efb13f847 --- /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 -- cgit v1.2.3-18-g5258 From da18396f949ecaa45007d3aeb1b81bd6da092811 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Feb 2013 15:44:07 +0000 Subject: ASoC: core: Allow digital mute for capture Help avoid noise from the power up of the capture path propagating through into the start of the recording (especially noise caused by the ramp of microphone biases) by keeping the capture muted until after we've finished powering things up with DAPM in the same manner we do for playback. This allows us to take advantage of soft mute support in the hardware more effectively and is more consistent. The core code using the existing digital mute operation is updated to take advantage of this. Some additional cases in the soc-pcm code and suspend will need separate handling but these are less practically relevant than the main runtime stream start/stop case. Rather than refactor the digital mute function in every single driver a new operation is added for drivers taking advantage of this functionality, the old operation should be phased out over time. Signed-off-by: Mark Brown Acked-by Vinod Koul Acked-by: Liam Girdwood --- include/sound/soc-dai.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3953cea0ecf..a680f23a04f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -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. -- cgit v1.2.3-18-g5258 From 9727b490e543de956b8ba356e2d5499097d0b7a2 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Thu, 14 Feb 2013 16:52:51 +0530 Subject: ALSA: compress: add support for gapless playback this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 8 ++++++++ include/uapi/sound/compress_offload.h | 31 ++++++++++++++++++++++++++++++- 2 files changed, 38 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912abacdf..ff6c74153fa 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/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a43fed..d630163b9a2 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,27 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +/** + * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the + * end of the track + * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the + * beginning of the track + */ +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +166,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +177,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif -- cgit v1.2.3-18-g5258 From ef5c2eba2412596f1a022c11caf74428bffd9abe Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 14 Feb 2013 12:02:51 +0000 Subject: ASoC: codecs: Add da7213 codec This patch adds support for the Dialog DA7213 audio codec. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7213.h | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 include/sound/da7213.h (limited to 'include') diff --git a/include/sound/da7213.h b/include/sound/da7213.h new file mode 100644 index 00000000000..673f5c39cbf --- /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 + * + * 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 */ -- cgit v1.2.3-18-g5258 From b531f81b0d70ffbe8d70500512483227cc532608 Mon Sep 17 00:00:00 2001 From: Pawel Moll Date: Thu, 21 Feb 2013 01:55:50 +0000 Subject: ALSA: usb: Fix Processing Unit Descriptor parsers Commit 99fc86450c439039d2ef88d06b222fd51a779176 "ALSA: usb-mixer: parse descriptors with structs" introduced a set of useful parsers for descriptors. Unfortunately the parses for the Processing Unit Descriptor came with a very subtle bug... Functions uac_processing_unit_iProcessing() and uac_processing_unit_specific() were indexing the baSourceID array forgetting the fields before the iProcessing and process-specific descriptors. The problem was observed with Sound Blaster Extigy mixer, where nNrModes in Up/Down-mix Processing Unit Descriptor was accessed at offset 10 of the descriptor (value 0) instead of offset 15 (value 7). In result the resulting control had interesting limit values: Simple mixer control 'Channel Routing Mode Select',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - -1 Mono: -1 [100%] Fixed by starting from the bmControls, which was calculated correctly, instead of baSourceID. Now the mentioned control is fine: Simple mixer control 'Channel Routing Mode Select',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 6 Mono: 0 [0%] Signed-off-by: Pawel Moll Cc: Signed-off-by: Takashi Iwai --- include/uapi/linux/usb/audio.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index ac90037894d..d2314be4f0c 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -384,14 +384,16 @@ static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_de int protocol) { __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); - return desc->baSourceID[desc->bNrInPins + control_size]; + return *(uac_processing_unit_bmControls(desc, protocol) + + control_size); } static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc, int protocol) { __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); - return &desc->baSourceID[desc->bNrInPins + control_size + 1]; + return uac_processing_unit_bmControls(desc, protocol) + + control_size + 1; } /* 4.5.2 Class-Specific AS Interface Descriptor */ -- cgit v1.2.3-18-g5258