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 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; -- cgit v1.2.3 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 --- Documentation/devicetree/bindings/sound/cs4271.txt | 12 ++++++++ include/sound/cs4271.h | 15 ++++++++++ sound/soc/codecs/cs4271.c | 34 ++++++++++++++++++++++ 3 files changed, 61 insertions(+) (limited to 'include') diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index a850fb9c88ea..e2cd1d7539e5 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -20,6 +20,18 @@ Optional properties: !RESET pin - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag is enabled. + - cirrus,enable-soft-reset: + 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 this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. Examples: 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/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac8742a1f25a..2415a4118dbd 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -167,6 +167,8 @@ struct cs4271_private { int gpio_nreset; /* GPIO that disable serial bus, if any */ int gpio_disable; + /* enable soft reset workaround */ + bool enable_soft_reset; }; /* @@ -325,6 +327,33 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, int i, ret; unsigned int ratio, val; + if (cs4271->enable_soft_reset) { + /* + * Put the codec in soft reset and back again in case it's not + * currently streaming data. This way of bringing the codec in + * sync to the current clocks is not explicitly documented in + * the data sheet, but it seems to work fine, and in contrast + * to a read hardware reset, we don't have to sync back all + * registers every time. + */ + + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !dai->capture_active) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + !dai->playback_active)) { + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); + if (ret < 0) + return ret; + + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + } + } + cs4271->rate = params_rate(params); /* Configure DAC */ @@ -484,6 +513,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; + + if (of_get_property(codec->dev->of_node, + "cirrus,enable-soft-reset", NULL)) + cs4271->enable_soft_reset = true; } #endif @@ -492,6 +525,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; + cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } if (gpio_nreset >= 0) -- cgit v1.2.3 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 ------------- sound/soc/sh/fsi.c | 131 +++++-------------------------------------------- 2 files changed, 12 insertions(+), 152 deletions(-) (limited to 'include') diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index cc1c919c6436..66285e1e340e 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 { diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a606d0f93d1c..5cb1332e0438 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -131,8 +131,6 @@ #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -typedef int (*set_rate_func)(struct device *dev, int rate, int enable); - /* * bus options * @@ -244,8 +242,7 @@ struct fsi_clk { struct clk *ick; struct clk *div; int (*set_rate)(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate); + struct fsi_priv *fsi); unsigned long rate; unsigned int count; @@ -270,8 +267,6 @@ struct fsi_priv { int enable_stream:1; int bit_clk_inv:1; int lr_clk_inv:1; - - long rate; }; struct fsi_stream_handler { @@ -431,14 +426,6 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return fsi_get_priv_frm_dai(fsi_get_dai(substream)); } -static set_rate_func fsi_get_info_set_rate(struct fsi_priv *fsi) -{ - if (!fsi->info) - return NULL; - - return fsi->info->set_rate; -} - static u32 fsi_get_info_flags(struct fsi_priv *fsi) { if (!fsi->info) @@ -757,8 +744,7 @@ static int fsi_clk_init(struct device *dev, int ick, int div, int (*set_rate)(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate)) + struct fsi_priv *fsi)) { struct fsi_clk *clock = &fsi->clock; int is_porta = fsi_is_port_a(fsi); @@ -829,8 +815,7 @@ static int fsi_clk_is_valid(struct fsi_priv *fsi) } static int fsi_clk_enable(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate) + struct fsi_priv *fsi) { struct fsi_clk *clock = &fsi->clock; int ret = -EINVAL; @@ -839,7 +824,7 @@ static int fsi_clk_enable(struct device *dev, return ret; if (0 == clock->count) { - ret = clock->set_rate(dev, fsi, rate); + ret = clock->set_rate(dev, fsi); if (ret < 0) { fsi_clk_invalid(fsi); return ret; @@ -946,11 +931,11 @@ static int fsi_clk_set_ackbpf(struct device *dev, } static int fsi_clk_set_rate_external(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate) + struct fsi_priv *fsi) { struct clk *xck = fsi->clock.xck; struct clk *ick = fsi->clock.ick; + unsigned long rate = fsi->clock.rate; unsigned long xrate; int ackmd, bpfmd; int ret = 0; @@ -978,11 +963,11 @@ static int fsi_clk_set_rate_external(struct device *dev, } static int fsi_clk_set_rate_cpg(struct device *dev, - struct fsi_priv *fsi, - unsigned long rate) + struct fsi_priv *fsi) { struct clk *ick = fsi->clock.ick; struct clk *div = fsi->clock.div; + unsigned long rate = fsi->clock.rate; unsigned long target = 0; /* 12288000 or 11289600 */ unsigned long actual, cout; unsigned long diff, min; @@ -1063,85 +1048,6 @@ static int fsi_clk_set_rate_cpg(struct device *dev, return ret; } -static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, - long rate, int enable) -{ - set_rate_func set_rate = fsi_get_info_set_rate(fsi); - int ret; - - /* - * CAUTION - * - * set_rate will be deleted - */ - if (!set_rate) { - if (enable) - return fsi_clk_enable(dev, fsi, rate); - else - return fsi_clk_disable(dev, fsi); - } - - ret = set_rate(dev, rate, enable); - if (ret < 0) /* error */ - return ret; - - if (!enable) - return 0; - - if (ret > 0) { - u32 data = 0; - - switch (ret & SH_FSI_ACKMD_MASK) { - default: - /* FALL THROUGH */ - case SH_FSI_ACKMD_512: - data |= (0x0 << 12); - break; - case SH_FSI_ACKMD_256: - data |= (0x1 << 12); - break; - case SH_FSI_ACKMD_128: - data |= (0x2 << 12); - break; - case SH_FSI_ACKMD_64: - data |= (0x3 << 12); - break; - case SH_FSI_ACKMD_32: - data |= (0x4 << 12); - break; - } - - switch (ret & SH_FSI_BPFMD_MASK) { - default: - /* FALL THROUGH */ - case SH_FSI_BPFMD_32: - data |= (0x0 << 8); - break; - case SH_FSI_BPFMD_64: - data |= (0x1 << 8); - break; - case SH_FSI_BPFMD_128: - data |= (0x2 << 8); - break; - case SH_FSI_BPFMD_256: - data |= (0x3 << 8); - break; - case SH_FSI_BPFMD_512: - data |= (0x4 << 8); - break; - case SH_FSI_BPFMD_16: - data |= (0x7 << 8); - break; - } - - fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data); - udelay(10); - ret = 0; - } - - return ret; -} - /* * pio data transfer handler */ @@ -1698,7 +1604,7 @@ static int fsi_hw_startup(struct fsi_priv *fsi, /* start master clock */ if (fsi_is_clk_master(fsi)) - return fsi_set_master_clk(dev, fsi, fsi->rate, 1); + return fsi_clk_enable(dev, fsi); return 0; } @@ -1708,7 +1614,7 @@ static int fsi_hw_shutdown(struct fsi_priv *fsi, { /* stop master clock */ if (fsi_is_clk_master(fsi)) - return fsi_set_master_clk(dev, fsi, fsi->rate, 0); + return fsi_clk_disable(dev, fsi); return 0; } @@ -1719,7 +1625,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, struct fsi_priv *fsi = fsi_get_priv(substream); fsi_clk_invalid(fsi); - fsi->rate = 0; return 0; } @@ -1730,7 +1635,6 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct fsi_priv *fsi = fsi_get_priv(substream); fsi_clk_invalid(fsi); - fsi->rate = 0; } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -1795,7 +1699,6 @@ static int fsi_set_fmt_spdif(struct fsi_priv *fsi) static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); - set_rate_func set_rate = fsi_get_info_set_rate(fsi); int ret; /* set master/slave audio interface */ @@ -1831,14 +1734,6 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } if (fsi_is_clk_master(fsi)) { - /* - * CAUTION - * - * set_rate will be deleted - */ - if (set_rate) - dev_warn(dai->dev, "set_rate will be removed soon\n"); - if (fsi->clk_cpg) fsi_clk_init(dai->dev, fsi, 0, 1, 1, fsi_clk_set_rate_cpg); @@ -1862,10 +1757,8 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); - if (fsi_is_clk_master(fsi)) { - fsi->rate = params_rate(params); - fsi_clk_valid(fsi, fsi->rate); - } + if (fsi_is_clk_master(fsi)) + fsi_clk_valid(fsi, params_rate(params)); return 0; } -- cgit v1.2.3 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 --- arch/arm/mach-shmobile/board-ap4evb.c | 11 +++-------- arch/arm/mach-shmobile/board-mackerel.c | 11 ++++------- arch/sh/boards/mach-ecovec24/setup.c | 12 ++---------- arch/sh/boards/mach-se/7724/setup.c | 12 ++---------- include/sound/sh_fsi.h | 7 ------- sound/soc/sh/fsi.c | 25 ------------------------- 6 files changed, 11 insertions(+), 67 deletions(-) (limited to 'include') diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 99ef190d0909..4c979039d97e 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -657,14 +657,8 @@ static struct platform_device lcdc_device = { /* FSI */ #define IRQ_FSI evt2irq(0x1840) static struct sh_fsi_platform_info fsi_info = { - .port_a = { - .flags = SH_FSI_BRS_INV, - }, .port_b = { - .flags = SH_FSI_BRS_INV | - SH_FSI_BRM_INV | - SH_FSI_LRS_INV | - SH_FSI_CLK_CPG | + .flags = SH_FSI_CLK_CPG | SH_FSI_FMT_SPDIF, }, }; @@ -816,7 +810,8 @@ static struct platform_device lcdc1_device = { }; static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, + .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct asoc_simple_card_info fsi2_hdmi_info = { diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index 2fed62f66045..b5d210b4264c 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -503,7 +503,8 @@ static struct platform_device hdmi_lcdc_device = { }; static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, + .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, }; static struct asoc_simple_card_info fsi2_hdmi_info = { @@ -858,16 +859,12 @@ static struct platform_device leds_device = { #define IRQ_FSI evt2irq(0x1840) static struct sh_fsi_platform_info fsi_info = { .port_a = { - .flags = SH_FSI_BRS_INV, .tx_id = SHDMA_SLAVE_FSIA_TX, .rx_id = SHDMA_SLAVE_FSIA_RX, }, .port_b = { - .flags = SH_FSI_BRS_INV | - SH_FSI_BRM_INV | - SH_FSI_LRS_INV | - SH_FSI_CLK_CPG | - SH_FSI_FMT_SPDIF, + .flags = SH_FSI_CLK_CPG | + SH_FSI_FMT_SPDIF, } }; diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 3fede4556c91..8ebe4c7a766b 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -877,12 +877,6 @@ static struct platform_device camera_devices[] = { }; /* FSI */ -static struct sh_fsi_platform_info fsi_info = { - .port_b = { - .flags = SH_FSI_BRS_INV, - }, -}; - static struct resource fsi_resources[] = { [0] = { .name = "FSI", @@ -901,15 +895,13 @@ static struct platform_device fsi_device = { .id = 0, .num_resources = ARRAY_SIZE(fsi_resources), .resource = fsi_resources, - .dev = { - .platform_data = &fsi_info, - }, }; static struct asoc_simple_dai_init_info fsi_da7210_init_info = { .fmt = SND_SOC_DAIFMT_I2S, .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, + .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_IB_NF, }; static struct asoc_simple_card_info fsi_da7210_info = { diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 35f6efa3ac0e..975608f5e805 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -279,12 +279,6 @@ static struct platform_device ceu1_device = { /* FSI */ /* change J20, J21, J22 pin to 1-2 connection to use slave mode */ -static struct sh_fsi_platform_info fsi_info = { - .port_a = { - .flags = SH_FSI_BRS_INV, - }, -}; - static struct resource fsi_resources[] = { [0] = { .name = "FSI", @@ -303,15 +297,13 @@ static struct platform_device fsi_device = { .id = 0, .num_resources = ARRAY_SIZE(fsi_resources), .resource = fsi_resources, - .dev = { - .platform_data = &fsi_info, - }, }; static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = { .fmt = SND_SOC_DAIFMT_LEFT_J, .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, + .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_IB_NF, .sysclk = 11289600, }; diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 66285e1e340e..43ac28581920 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) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5cb1332e0438..f14c611b38c6 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -251,7 +251,6 @@ struct fsi_clk { struct fsi_priv { void __iomem *base; struct fsi_master *master; - struct sh_fsi_port_info *info; struct fsi_stream playback; struct fsi_stream capture; @@ -426,14 +425,6 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return fsi_get_priv_frm_dai(fsi_get_dai(substream)); } -static u32 fsi_get_info_flags(struct fsi_priv *fsi) -{ - if (!fsi->info) - return 0; - - return fsi->info->flags; -} - static u32 fsi_get_port_shift(struct fsi_priv *fsi, struct fsi_stream *io) { int is_play = fsi_stream_is_play(fsi, io); @@ -1543,7 +1534,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { - u32 flags = fsi_get_info_flags(fsi); u32 data = 0; /* clock setting */ @@ -1560,19 +1550,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, data |= (1 << 4); if (fsi_is_clk_master(fsi)) data <<= 8; - /* FIXME - * - * SH_FSI_xxx_INV style will be removed - */ - if (SH_FSI_LRM_INV & flags) - data |= 1 << 12; - if (SH_FSI_BRM_INV & flags) - data |= 1 << 8; - if (SH_FSI_LRS_INV & flags) - data |= 1 << 4; - if (SH_FSI_BRS_INV & flags) - data |= 1 << 0; - fsi_reg_write(fsi, CKG2, data); /* spdif ? */ @@ -1988,7 +1965,6 @@ static int fsi_probe(struct platform_device *pdev) fsi = &master->fsia; fsi->base = master->base; fsi->master = master; - fsi->info = pinfo; fsi_port_info_init(fsi, pinfo); fsi_handler_init(fsi, pinfo); ret = fsi_stream_probe(fsi, &pdev->dev); @@ -2002,7 +1978,6 @@ static int fsi_probe(struct platform_device *pdev) fsi = &master->fsib; fsi->base = master->base + 0x40; fsi->master = master; - fsi->info = pinfo; fsi_port_info_init(fsi, pinfo); fsi_handler_init(fsi, pinfo); ret = fsi_stream_probe(fsi, &pdev->dev); -- cgit v1.2.3 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 43ac28581920..7a9710b4b799 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 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 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 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 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/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 365d9d27a321..752675da0658 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include #include #include -#include #include #include #include @@ -662,43 +661,32 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- cgit v1.2.3 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 --- arch/arm/mach-shmobile/board-ap4evb.c | 35 ++++++++-------- arch/arm/mach-shmobile/board-armadillo800eva.c | 34 +++++++-------- arch/arm/mach-shmobile/board-kzm9g.c | 20 ++++----- arch/arm/mach-shmobile/board-mackerel.c | 35 ++++++++-------- arch/sh/boards/mach-ecovec24/setup.c | 19 ++++----- arch/sh/boards/mach-se/7724/setup.c | 21 +++++----- include/sound/simple_card.h | 12 +++--- sound/soc/generic/simple-card.c | 58 +++++++++++++------------- 8 files changed, 116 insertions(+), 118 deletions(-) (limited to 'include') diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 4c979039d97e..08294fa9e0d4 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -686,21 +686,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { @@ -809,19 +809,18 @@ static struct platform_device lcdc1_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 5353adf6b828..0679ca6bf1f6 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -806,21 +806,21 @@ static struct platform_device fsi_device = { }; /* FSI-WM8978 */ -static struct asoc_simple_dai_init_info fsi_wm8978_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 12288000, -}; - static struct asoc_simple_card_info fsi_wm8978_info = { .name = "wm8978", .card = "FSI2A-WM8978", - .cpu_dai = "fsia-dai", .codec = "wm8978.0-001a", .platform = "sh_fsi2", - .codec_dai = "wm8978-hifi", - .init = &fsi_wm8978_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "wm8978-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, + .sysclk = 12288000, + }, }; static struct platform_device fsi_wm8978_device = { @@ -832,18 +832,18 @@ static struct platform_device fsi_wm8978_device = { }; /* FSI-HDMI */ -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index c02448d6847f..f41b71e8df3e 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -525,21 +525,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4648_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4648_info = { .name = "AK4648", .card = "FSI2A-AK4648", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4648_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4648_device = { diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index b5d210b4264c..3fd716dae405 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -502,19 +502,18 @@ static struct platform_device hdmi_lcdc_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { @@ -893,21 +892,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 8ebe4c7a766b..065e9600fae6 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -897,21 +897,20 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi_da7210_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi_da7210_info = { .name = "DA7210", .card = "FSIB-DA7210", - .cpu_dai = "fsib-dai", .codec = "da7210.0-001a", .platform = "sh_fsi.0", - .codec_dai = "da7210-hifi", - .init = &fsi_da7210_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "da7210-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, }; static struct platform_device fsi_da7210_device = { diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 975608f5e805..4010e63e82d8 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -299,22 +299,21 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi_ak4642_info = { .name = "AK4642", .card = "FSIA-AK4642", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4642_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4642_device = { 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 -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/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bc050ec8680a..6cf8355a8542 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,33 +16,38 @@ #define asoc_simple_get_card_info(p) \ container_of(p->dai_link, struct asoc_simple_card_info, snd_link) +static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, + struct asoc_simple_dai *set, + unsigned int daifmt) +{ + int ret = 0; + + daifmt |= set->fmt; + + if (!ret && daifmt) + ret = snd_soc_dai_set_fmt(dai, daifmt); + + if (!ret && set->sysclk) + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + + return ret; +} + static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); - struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; - unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + unsigned int daifmt = info->daifmt; int ret; - if (codec_daifmt) { - ret = snd_soc_dai_set_fmt(codec, codec_daifmt); - if (ret < 0) - return ret; - } - - if (iinfo->sysclk) { - ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + if (ret < 0) + return ret; - if (cpu_daifmt) { - ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + if (ret < 0) + return ret; return 0; } @@ -59,10 +64,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!cinfo->name || !cinfo->card || - !cinfo->cpu_dai || !cinfo->codec || !cinfo->platform || - !cinfo->codec_dai) { + !cinfo->cpu_dai.name || + !cinfo->codec_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -72,14 +77,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai; - - /* enable snd_link.init if cinfo has settings */ - if (cinfo->init) - cinfo->snd_link.init = asoc_simple_card_dai_init; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; + cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card -- cgit v1.2.3 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 769e27c774a3..bedf3dabdbd3 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 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 3953cea0ecfb..90dc00434da8 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 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 +++++++++ sound/soc/codecs/arizona.c | 14 ++++++++++++-- 2 files changed, 21 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/linux/mfd/arizona/pdata.h b/include/linux/mfd/arizona/pdata.h index 8b1d1daaae16..ec3e2a2a6d77 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; diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 845d25630ba2..d855a6c098d4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -762,18 +762,28 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; int base = dai->driver->base; const int *rates; int i, ret; - int bclk, lrclk, wl, frame; + int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; + int bclk, lrclk, wl, frame, bclk_target; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; else rates = &arizona_48k_bclk_rates[0]; + bclk_target = snd_soc_params_to_bclk(params); + if (chan_limit && chan_limit < params_channels(params)) { + arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit); + bclk_target /= params_channels(params); + bclk_target *= chan_limit; + } + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { - if (rates[i] >= snd_soc_params_to_bclk(params) && + if (rates[i] >= bclk_target && rates[i] % params_rate(params) == 0) { bclk = i; break; -- cgit v1.2.3 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 +++++++++--- sound/drivers/vx/vx_core.c | 3 +-- sound/pci/atiixp.c | 5 +++-- sound/pci/hda/hda_auto_parser.c | 2 -- sound/pci/hda/hda_generic.c | 2 -- sound/pci/intel8x0.c | 10 ++++++---- sound/pci/maestro3.c | 10 ++++++---- sound/pci/nm256/nm256.c | 3 ++- sound/pci/pcxhr/pcxhr_core.c | 3 +-- sound/pci/via82xx.c | 2 +- sound/usb/pcm.c | 2 -- 11 files changed, 29 insertions(+), 25 deletions(-) (limited to 'include') 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/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index de5055a3b0d0..c39961c11401 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -52,7 +52,6 @@ MODULE_LICENSE("GPL"); int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int time) { unsigned long end_time = jiffies + (time * HZ + 999) / 1000; -#ifdef CONFIG_SND_DEBUG static char *reg_names[VX_REG_MAX] = { "ICR", "CVR", "ISR", "IVR", "RXH", "RXM", "RXL", "DMA", "CDSP", "RFREQ", "RUER/V2", "DATA", "MEMIRQ", @@ -60,7 +59,7 @@ int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int t "MIC3", "INTCSR", "CNTRL", "GPIOC", "LOFREQ", "HIFREQ", "CSUER", "RUER" }; -#endif + do { if ((snd_vx_inb(chip, reg) & mask) == bit) return 0; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index a67743183aaf..6e78c6789858 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -567,8 +567,9 @@ static int ac97_probing_bugs(struct pci_dev *pci) q = snd_pci_quirk_lookup(pci, atiixp_quirks); if (q) { - snd_printdd(KERN_INFO "Atiixp quirk for %s. " - "Forcing codec %d\n", q->name, q->value); + snd_printdd(KERN_INFO + "Atiixp quirk for %s. Forcing codec %d\n", + snd_pci_quirk_name(q), q->value); return q->value; } /* this hardware doesn't need workarounds. Probe for codec */ diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 96a05c4c28e1..a3ea76a4c9d2 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -698,9 +698,7 @@ static void set_pin_targets(struct hda_codec *codec, static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) { -#ifdef CONFIG_SND_DEBUG_VERBOSE const char *modelname = codec->fixup_name; -#endif while (id >= 0) { const struct hda_fixup *fix = codec->fixup_list + id; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 19d014a6a40e..c4ba3066a013 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1579,9 +1579,7 @@ static void debug_show_configs(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct hda_gen_spec *spec = codec->spec; -#ifdef CONFIG_SND_DEBUG_VERBOSE static const char * const lo_type[3] = { "LO", "SP", "HP" }; -#endif int i; debug_badness("multi_outs = %x/%x/%x/%x : %x/%x/%x/%x (type %s)\n", diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 3b9be752f3e2..b8fe40531b9c 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3266,11 +3266,13 @@ static int check_default_spdif_aclink(struct pci_dev *pci) w = snd_pci_quirk_lookup(pci, spdif_aclink_defaults); if (w) { if (w->value) - snd_printdd(KERN_INFO "intel8x0: Using SPDIF over " - "AC-Link for %s\n", w->name); + snd_printdd(KERN_INFO + "intel8x0: Using SPDIF over AC-Link for %s\n", + snd_pci_quirk_name(w)); else - snd_printdd(KERN_INFO "intel8x0: Using integrated " - "SPDIF DMA for %s\n", w->name); + snd_printdd(KERN_INFO + "intel8x0: Using integrated SPDIF DMA for %s\n", + snd_pci_quirk_name(w)); return w->value; } return 0; diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 9387533f70dc..c76ac1411210 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2586,8 +2586,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, else { quirk = snd_pci_quirk_lookup(pci, m3_amp_quirk_list); if (quirk) { - snd_printdd(KERN_INFO "maestro3: set amp-gpio " - "for '%s'\n", quirk->name); + snd_printdd(KERN_INFO + "maestro3: set amp-gpio for '%s'\n", + snd_pci_quirk_name(quirk)); chip->amp_gpio = quirk->value; } else if (chip->allegro_flag) chip->amp_gpio = GPO_EXT_AMP_ALLEGRO; @@ -2597,8 +2598,9 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, quirk = snd_pci_quirk_lookup(pci, m3_irda_quirk_list); if (quirk) { - snd_printdd(KERN_INFO "maestro3: enabled irda workaround " - "for '%s'\n", quirk->name); + snd_printdd(KERN_INFO + "maestro3: enabled irda workaround for '%s'\n", + snd_pci_quirk_name(quirk)); chip->irda_workaround = 1; } quirk = snd_pci_quirk_lookup(pci, m3_hv_quirk_list); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 563a193e36a3..6febedb05936 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1660,7 +1660,8 @@ static int snd_nm256_probe(struct pci_dev *pci, q = snd_pci_quirk_lookup(pci, nm256_quirks); if (q) { - snd_printdd(KERN_INFO "nm256: Enabled quirk for %s.\n", q->name); + snd_printdd(KERN_INFO "nm256: Enabled quirk for %s.\n", + snd_pci_quirk_name(q)); switch (q->value) { case NM_BLACKLISTED: printk(KERN_INFO "nm256: The device is blacklisted. " diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index b33db1e006e7..37b431b9b69d 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1012,13 +1012,12 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err, enum pcxhr_async_err_src err_src, int pipe, int is_capture) { -#ifdef CONFIG_SND_DEBUG_VERBOSE static char* err_src_name[] = { [PCXHR_ERR_PIPE] = "Pipe", [PCXHR_ERR_STREAM] = "Stream", [PCXHR_ERR_AUDIO] = "Audio" }; -#endif + if (err & 0xfff) err &= 0xfff; else diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 6442f611a07b..d756a3562706 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2517,7 +2517,7 @@ static int check_dxs_list(struct pci_dev *pci, int revision) w = snd_pci_quirk_lookup(pci, dxs_whitelist); if (w) { snd_printdd(KERN_INFO "via82xx: DXS white list for %s found\n", - w->name); + snd_pci_quirk_name(w)); return w->value; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b839b60f9858..81f70a719bb9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1179,9 +1179,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, if (!subs->txfr_quirk) bytes = frames * stride; if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE int oldbytes = bytes; -#endif bytes = frames * stride; snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", oldbytes, bytes); -- cgit v1.2.3 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 + sound/soc/soc-core.c | 115 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 117 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index bc56738cb109..571562340e5a 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 diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2370063b5824..9d07dc03a1d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4208,6 +4208,121 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_daifmt(struct device_node *np, + const char *prefix) +{ + int ret, i; + char prop[128]; + unsigned int format = 0; + int bit, frame; + const char *str; + struct { + char *name; + unsigned int val; + } of_fmt_table[] = { + { "i2s", SND_SOC_DAIFMT_I2S }, + { "right_j", SND_SOC_DAIFMT_RIGHT_J }, + { "left_j", SND_SOC_DAIFMT_LEFT_J }, + { "dsp_a", SND_SOC_DAIFMT_DSP_A }, + { "dsp_b", SND_SOC_DAIFMT_DSP_B }, + { "ac97", SND_SOC_DAIFMT_AC97 }, + { "pdm", SND_SOC_DAIFMT_PDM}, + { "msb", SND_SOC_DAIFMT_MSB }, + { "lsb", SND_SOC_DAIFMT_LSB }, + }, of_clock_table[] = { + { "continuous", SND_SOC_DAIFMT_CONT }, + { "gated", SND_SOC_DAIFMT_GATED }, + }; + + if (!prefix) + prefix = ""; + + /* + * check "[prefix]format = xxx" + * SND_SOC_DAIFMT_FORMAT_MASK area + */ + snprintf(prop, sizeof(prop), "%sformat", prefix); + ret = of_property_read_string(np, prop, &str); + if (ret == 0) { + for (i = 0; i < ARRAY_SIZE(of_fmt_table); i++) { + if (strcmp(str, of_fmt_table[i].name) == 0) { + format |= of_fmt_table[i].val; + break; + } + } + } + + /* + * check "[prefix]clock-gating = xxx" + * SND_SOC_DAIFMT_CLOCK_MASK area + */ + snprintf(prop, sizeof(prop), "%sclock-gating", prefix); + ret = of_property_read_string(np, prop, &str); + if (ret == 0) { + for (i = 0; i < ARRAY_SIZE(of_clock_table); i++) { + if (strcmp(str, of_clock_table[i].name) == 0) { + format |= of_clock_table[i].val; + break; + } + } + } + + /* + * check "[prefix]bitclock-inversion" + * check "[prefix]frame-inversion" + * SND_SOC_DAIFMT_INV_MASK area + */ + snprintf(prop, sizeof(prop), "%sbitclock-inversion", prefix); + bit = !!of_get_property(np, prop, NULL); + + snprintf(prop, sizeof(prop), "%sframe-inversion", prefix); + frame = !!of_get_property(np, prop, NULL); + + switch ((bit << 4) + frame) { + case 0x11: + format |= SND_SOC_DAIFMT_IB_IF; + break; + case 0x10: + format |= SND_SOC_DAIFMT_IB_NF; + break; + case 0x01: + format |= SND_SOC_DAIFMT_NB_IF; + break; + default: + /* SND_SOC_DAIFMT_NB_NF is default */ + break; + } + + /* + * check "[prefix]bitclock-master" + * check "[prefix]frame-master" + * SND_SOC_DAIFMT_MASTER_MASK area + */ + snprintf(prop, sizeof(prop), "%sbitclock-master", prefix); + bit = !!of_get_property(np, prop, NULL); + + snprintf(prop, sizeof(prop), "%sframe-master", prefix); + frame = !!of_get_property(np, prop, NULL); + + switch ((bit << 4) + frame) { + case 0x11: + format |= SND_SOC_DAIFMT_CBM_CFM; + break; + case 0x10: + format |= SND_SOC_DAIFMT_CBM_CFS; + break; + case 0x01: + format |= SND_SOC_DAIFMT_CBS_CFM; + break; + default: + format |= SND_SOC_DAIFMT_CBS_CFS; + break; + } + + return format; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- cgit v1.2.3 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 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 -- cgit v1.2.3 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 3953cea0ecfb..4dbd3e78eb87 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 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 --- .../devicetree/bindings/sound/tlv320aic3x.txt | 6 ++ include/sound/tlv320aic3x.h | 10 +++ sound/soc/codecs/tlv320aic3x.c | 83 ++++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.h | 4 ++ sound/soc/davinci/davinci-evm.c | 6 +- sound/soc/omap/n810.c | 4 +- sound/soc/omap/rx51.c | 8 +-- 7 files changed, 106 insertions(+), 15 deletions(-) (limited to 'include') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index e7b98f41fa5f..f47c3f589fd0 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -11,6 +11,12 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality +- ai3x-micbias-vg - MicBias Voltage required. + 1 - MICBIAS output is powered to 2.0V, + 2 - MICBIAS output is powered to 2.5V, + 3 - MICBIAS output is connected to AVDD, + If this node is not mentioned or if the value is incorrect, then MicBias + is powered down. Example: 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/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5708a973a776..ba82ba2a7133 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -85,6 +85,9 @@ struct aic3x_priv { #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 u16 model; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; /* @@ -195,6 +198,37 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +/* + * mic bias power on/off share the same register bits with + * output voltage of mic bias. when power on mic bias, we + * need reclaim it to voltage value. + * 0x0 = Powered off + * 0x1 = MICBIAS output is powered to 2.0V, + * 0x2 = MICBIAS output is powered to 2.5V + * 0x3 = MICBIAS output is connected to AVDD + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, 0); + break; + } + return 0; +} + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -596,12 +630,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), /* Mic Bias */ - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", - MICBIAS_CTRL, 6, 3, 1, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", - MICBIAS_CTRL, 6, 3, 2, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", - MICBIAS_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, @@ -1386,6 +1417,24 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + /* set mic bias voltage */ + switch (aic3x->micbias_vg) { + case AIC3X_MICBIAS_2_0V: + case AIC3X_MICBIAS_2_5V: + case AIC3X_MICBIAS_AVDDV: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + (aic3x->micbias_vg) << MICBIAS_LEVEL_SHIFT); + break; + case AIC3X_MICBIAS_OFF: + /* + * noting to do. target won't enter here. This is just to avoid + * compile time warning "warning: enumeration value + * 'AIC3X_MICBIAS_OFF' not handled in switch" + */ + break; + } + aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); @@ -1461,6 +1510,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; int ret; + u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1474,6 +1524,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + aic3x->micbias_vg = pdata->micbias_vg; } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); @@ -1493,6 +1544,26 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->setup = ai3x_setup; } + if (!of_property_read_u32(np, "ai3x-micbias-vg", &value)) { + switch (value) { + case 1 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_0V; + break; + case 2 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_5V; + break; + case 3 : + aic3x->micbias_vg = AIC3X_MICBIAS_AVDDV; + break; + default : + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + dev_err(&i2c->dev, "Unsuitable MicBias voltage " + "found in DT\n"); + } + } else { + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + } + } else { aic3x->gpio_reset = -1; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6db3c41b0163..e521ac3ddde8 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -238,6 +238,10 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* MICBIAS Control Register */ +#define MICBIAS_LEVEL_SHIFT (6) +#define MICBIAS_LEVEL_MASK (3 << 6) + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index d55e6477bff0..484b22c5df5d 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -116,9 +116,9 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Line Out", NULL, "RLOUT"}, /* Mic connected to (MIC3L | MIC3R) */ - {"MIC3L", NULL, "Mic Bias 2V"}, - {"MIC3R", NULL, "Mic Bias 2V"}, - {"Mic Bias 2V", NULL, "Mic Jack"}, + {"MIC3L", NULL, "Mic Bias"}, + {"MIC3R", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Mic Jack"}, /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ {"LINE1L", NULL, "Line In"}, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 230b8c144848..ee7cd53aa3ee 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -230,8 +230,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "LLOUT"}, {"Ext Spk", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias 2V"}, - {"Mic Bias 2V", NULL, "DMic"}, + {"DMic Rate 64", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "DMic"}, }; static const char *spk_function[] = {"Off", "On"}; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index d921ddbe3ecb..3cd525748975 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -248,16 +248,16 @@ static const struct snd_soc_dapm_route audio_map[] = { {"FM Transmitter", NULL, "LLOUT"}, {"FM Transmitter", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias 2V"}, - {"Mic Bias 2V", NULL, "DMic"}, + {"DMic Rate 64", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "DMic"}, }; static const struct snd_soc_dapm_route audio_mapb[] = { {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, - {"LINE1L", NULL, "b Mic Bias 2.5V"}, - {"b Mic Bias 2.5V", NULL, "HS Mic"} + {"LINE1L", NULL, "b Mic Bias"}, + {"b Mic Bias", NULL, "HS Mic"} }; static const char *spk_function[] = {"Off", "On"}; -- cgit v1.2.3 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 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletion(-) (limited to 'include') 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 diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index fee1a18e309e..31d29c8f11bc 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2212,6 +2212,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2362,6 +2363,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- cgit v1.2.3 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 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98090.c | 2398 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98090.h | 1549 ++++++++++++++++++++++++++++ 5 files changed, 3982 insertions(+) create mode 100755 include/sound/max98090.h create mode 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h (limited to 'include') 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/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0e368d469764..3a847828932a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -267,6 +268,9 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98090 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aa5631220166..f6e8e36cceb7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o +snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -157,6 +158,7 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c new file mode 100755 index 000000000000..fc176044994d --- /dev/null +++ b/sound/soc/codecs/max98090.c @@ -0,0 +1,2398 @@ +/* + * max98090.c -- MAX98090 ALSA SoC Audio driver + * + * 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 version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "max98090.h" + +#include + +#define DEBUG +#define EXTMIC_METHOD +#define EXTMIC_METHOD_TEST + +/* Allows for sparsely populated register maps */ +static struct reg_default max98090_reg[] = { + { 0x00, 0x00 }, /* 00 Software Reset */ + { 0x03, 0x04 }, /* 03 Interrupt Masks */ + { 0x04, 0x00 }, /* 04 System Clock Quick */ + { 0x05, 0x00 }, /* 05 Sample Rate Quick */ + { 0x06, 0x00 }, /* 06 DAI Interface Quick */ + { 0x07, 0x00 }, /* 07 DAC Path Quick */ + { 0x08, 0x00 }, /* 08 Mic/Direct to ADC Quick */ + { 0x09, 0x00 }, /* 09 Line to ADC Quick */ + { 0x0A, 0x00 }, /* 0A Analog Mic Loop Quick */ + { 0x0B, 0x00 }, /* 0B Analog Line Loop Quick */ + { 0x0C, 0x00 }, /* 0C Reserved */ + { 0x0D, 0x00 }, /* 0D Input Config */ + { 0x0E, 0x1B }, /* 0E Line Input Level */ + { 0x0F, 0x00 }, /* 0F Line Config */ + + { 0x10, 0x14 }, /* 10 Mic1 Input Level */ + { 0x11, 0x14 }, /* 11 Mic2 Input Level */ + { 0x12, 0x00 }, /* 12 Mic Bias Voltage */ + { 0x13, 0x00 }, /* 13 Digital Mic Config */ + { 0x14, 0x00 }, /* 14 Digital Mic Mode */ + { 0x15, 0x00 }, /* 15 Left ADC Mixer */ + { 0x16, 0x00 }, /* 16 Right ADC Mixer */ + { 0x17, 0x03 }, /* 17 Left ADC Level */ + { 0x18, 0x03 }, /* 18 Right ADC Level */ + { 0x19, 0x00 }, /* 19 ADC Biquad Level */ + { 0x1A, 0x00 }, /* 1A ADC Sidetone */ + { 0x1B, 0x00 }, /* 1B System Clock */ + { 0x1C, 0x00 }, /* 1C Clock Mode */ + { 0x1D, 0x00 }, /* 1D Any Clock 1 */ + { 0x1E, 0x00 }, /* 1E Any Clock 2 */ + { 0x1F, 0x00 }, /* 1F Any Clock 3 */ + + { 0x20, 0x00 }, /* 20 Any Clock 4 */ + { 0x21, 0x00 }, /* 21 Master Mode */ + { 0x22, 0x00 }, /* 22 Interface Format */ + { 0x23, 0x00 }, /* 23 TDM Format 1*/ + { 0x24, 0x00 }, /* 24 TDM Format 2*/ + { 0x25, 0x00 }, /* 25 I/O Configuration */ + { 0x26, 0x80 }, /* 26 Filter Config */ + { 0x27, 0x00 }, /* 27 DAI Playback Level */ + { 0x28, 0x00 }, /* 28 EQ Playback Level */ + { 0x29, 0x00 }, /* 29 Left HP Mixer */ + { 0x2A, 0x00 }, /* 2A Right HP Mixer */ + { 0x2B, 0x00 }, /* 2B HP Control */ + { 0x2C, 0x1A }, /* 2C Left HP Volume */ + { 0x2D, 0x1A }, /* 2D Right HP Volume */ + { 0x2E, 0x00 }, /* 2E Left Spk Mixer */ + { 0x2F, 0x00 }, /* 2F Right Spk Mixer */ + + { 0x30, 0x00 }, /* 30 Spk Control */ + { 0x31, 0x2C }, /* 31 Left Spk Volume */ + { 0x32, 0x2C }, /* 32 Right Spk Volume */ + { 0x33, 0x00 }, /* 33 ALC Timing */ + { 0x34, 0x00 }, /* 34 ALC Compressor */ + { 0x35, 0x00 }, /* 35 ALC Expander */ + { 0x36, 0x00 }, /* 36 ALC Gain */ + { 0x37, 0x00 }, /* 37 Rcv/Line OutL Mixer */ + { 0x38, 0x00 }, /* 38 Rcv/Line OutL Control */ + { 0x39, 0x15 }, /* 39 Rcv/Line OutL Volume */ + { 0x3A, 0x00 }, /* 3A Line OutR Mixer */ + { 0x3B, 0x00 }, /* 3B Line OutR Control */ + { 0x3C, 0x15 }, /* 3C Line OutR Volume */ + { 0x3D, 0x00 }, /* 3D Jack Detect */ + { 0x3E, 0x00 }, /* 3E Input Enable */ + { 0x3F, 0x00 }, /* 3F Output Enable */ + + { 0x40, 0x00 }, /* 40 Level Control */ + { 0x41, 0x00 }, /* 41 DSP Filter Enable */ + { 0x42, 0x00 }, /* 42 Bias Control */ + { 0x43, 0x00 }, /* 43 DAC Control */ + { 0x44, 0x06 }, /* 44 ADC Control */ + { 0x45, 0x00 }, /* 45 Device Shutdown */ + { 0x46, 0x00 }, /* 46 Equalizer Band 1 Coefficient B0 */ + { 0x47, 0x00 }, /* 47 Equalizer Band 1 Coefficient B0 */ + { 0x48, 0x00 }, /* 48 Equalizer Band 1 Coefficient B0 */ + { 0x49, 0x00 }, /* 49 Equalizer Band 1 Coefficient B1 */ + { 0x4A, 0x00 }, /* 4A Equalizer Band 1 Coefficient B1 */ + { 0x4B, 0x00 }, /* 4B Equalizer Band 1 Coefficient B1 */ + { 0x4C, 0x00 }, /* 4C Equalizer Band 1 Coefficient B2 */ + { 0x4D, 0x00 }, /* 4D Equalizer Band 1 Coefficient B2 */ + { 0x4E, 0x00 }, /* 4E Equalizer Band 1 Coefficient B2 */ + { 0x4F, 0x00 }, /* 4F Equalizer Band 1 Coefficient A1 */ + + { 0x50, 0x00 }, /* 50 Equalizer Band 1 Coefficient A1 */ + { 0x51, 0x00 }, /* 51 Equalizer Band 1 Coefficient A1 */ + { 0x52, 0x00 }, /* 52 Equalizer Band 1 Coefficient A2 */ + { 0x53, 0x00 }, /* 53 Equalizer Band 1 Coefficient A2 */ + { 0x54, 0x00 }, /* 54 Equalizer Band 1 Coefficient A2 */ + { 0x55, 0x00 }, /* 55 Equalizer Band 2 Coefficient B0 */ + { 0x56, 0x00 }, /* 56 Equalizer Band 2 Coefficient B0 */ + { 0x57, 0x00 }, /* 57 Equalizer Band 2 Coefficient B0 */ + { 0x58, 0x00 }, /* 58 Equalizer Band 2 Coefficient B1 */ + { 0x59, 0x00 }, /* 59 Equalizer Band 2 Coefficient B1 */ + { 0x5A, 0x00 }, /* 5A Equalizer Band 2 Coefficient B1 */ + { 0x5B, 0x00 }, /* 5B Equalizer Band 2 Coefficient B2 */ + { 0x5C, 0x00 }, /* 5C Equalizer Band 2 Coefficient B2 */ + { 0x5D, 0x00 }, /* 5D Equalizer Band 2 Coefficient B2 */ + { 0x5E, 0x00 }, /* 5E Equalizer Band 2 Coefficient A1 */ + { 0x5F, 0x00 }, /* 5F Equalizer Band 2 Coefficient A1 */ + + { 0x60, 0x00 }, /* 60 Equalizer Band 2 Coefficient A1 */ + { 0x61, 0x00 }, /* 61 Equalizer Band 2 Coefficient A2 */ + { 0x62, 0x00 }, /* 62 Equalizer Band 2 Coefficient A2 */ + { 0x63, 0x00 }, /* 63 Equalizer Band 2 Coefficient A2 */ + { 0x64, 0x00 }, /* 64 Equalizer Band 3 Coefficient B0 */ + { 0x65, 0x00 }, /* 65 Equalizer Band 3 Coefficient B0 */ + { 0x66, 0x00 }, /* 66 Equalizer Band 3 Coefficient B0 */ + { 0x67, 0x00 }, /* 67 Equalizer Band 3 Coefficient B1 */ + { 0x68, 0x00 }, /* 68 Equalizer Band 3 Coefficient B1 */ + { 0x69, 0x00 }, /* 69 Equalizer Band 3 Coefficient B1 */ + { 0x6A, 0x00 }, /* 6A Equalizer Band 3 Coefficient B2 */ + { 0x6B, 0x00 }, /* 6B Equalizer Band 3 Coefficient B2 */ + { 0x6C, 0x00 }, /* 6C Equalizer Band 3 Coefficient B2 */ + { 0x6D, 0x00 }, /* 6D Equalizer Band 3 Coefficient A1 */ + { 0x6E, 0x00 }, /* 6E Equalizer Band 3 Coefficient A1 */ + { 0x6F, 0x00 }, /* 6F Equalizer Band 3 Coefficient A1 */ + + { 0x70, 0x00 }, /* 70 Equalizer Band 3 Coefficient A2 */ + { 0x71, 0x00 }, /* 71 Equalizer Band 3 Coefficient A2 */ + { 0x72, 0x00 }, /* 72 Equalizer Band 3 Coefficient A2 */ + { 0x73, 0x00 }, /* 73 Equalizer Band 4 Coefficient B0 */ + { 0x74, 0x00 }, /* 74 Equalizer Band 4 Coefficient B0 */ + { 0x75, 0x00 }, /* 75 Equalizer Band 4 Coefficient B0 */ + { 0x76, 0x00 }, /* 76 Equalizer Band 4 Coefficient B1 */ + { 0x77, 0x00 }, /* 77 Equalizer Band 4 Coefficient B1 */ + { 0x78, 0x00 }, /* 78 Equalizer Band 4 Coefficient B1 */ + { 0x79, 0x00 }, /* 79 Equalizer Band 4 Coefficient B2 */ + { 0x7A, 0x00 }, /* 7A Equalizer Band 4 Coefficient B2 */ + { 0x7B, 0x00 }, /* 7B Equalizer Band 4 Coefficient B2 */ + { 0x7C, 0x00 }, /* 7C Equalizer Band 4 Coefficient A1 */ + { 0x7D, 0x00 }, /* 7D Equalizer Band 4 Coefficient A1 */ + { 0x7E, 0x00 }, /* 7E Equalizer Band 4 Coefficient A1 */ + { 0x7F, 0x00 }, /* 7F Equalizer Band 4 Coefficient A2 */ + + { 0x80, 0x00 }, /* 80 Equalizer Band 4 Coefficient A2 */ + { 0x81, 0x00 }, /* 81 Equalizer Band 4 Coefficient A2 */ + { 0x82, 0x00 }, /* 82 Equalizer Band 5 Coefficient B0 */ + { 0x83, 0x00 }, /* 83 Equalizer Band 5 Coefficient B0 */ + { 0x84, 0x00 }, /* 84 Equalizer Band 5 Coefficient B0 */ + { 0x85, 0x00 }, /* 85 Equalizer Band 5 Coefficient B1 */ + { 0x86, 0x00 }, /* 86 Equalizer Band 5 Coefficient B1 */ + { 0x87, 0x00 }, /* 87 Equalizer Band 5 Coefficient B1 */ + { 0x88, 0x00 }, /* 88 Equalizer Band 5 Coefficient B2 */ + { 0x89, 0x00 }, /* 89 Equalizer Band 5 Coefficient B2 */ + { 0x8A, 0x00 }, /* 8A Equalizer Band 5 Coefficient B2 */ + { 0x8B, 0x00 }, /* 8B Equalizer Band 5 Coefficient A1 */ + { 0x8C, 0x00 }, /* 8C Equalizer Band 5 Coefficient A1 */ + { 0x8D, 0x00 }, /* 8D Equalizer Band 5 Coefficient A1 */ + { 0x8E, 0x00 }, /* 8E Equalizer Band 5 Coefficient A2 */ + { 0x8F, 0x00 }, /* 8F Equalizer Band 5 Coefficient A2 */ + + { 0x90, 0x00 }, /* 90 Equalizer Band 5 Coefficient A2 */ + { 0x91, 0x00 }, /* 91 Equalizer Band 6 Coefficient B0 */ + { 0x92, 0x00 }, /* 92 Equalizer Band 6 Coefficient B0 */ + { 0x93, 0x00 }, /* 93 Equalizer Band 6 Coefficient B0 */ + { 0x94, 0x00 }, /* 94 Equalizer Band 6 Coefficient B1 */ + { 0x95, 0x00 }, /* 95 Equalizer Band 6 Coefficient B1 */ + { 0x96, 0x00 }, /* 96 Equalizer Band 6 Coefficient B1 */ + { 0x97, 0x00 }, /* 97 Equalizer Band 6 Coefficient B2 */ + { 0x98, 0x00 }, /* 98 Equalizer Band 6 Coefficient B2 */ + { 0x99, 0x00 }, /* 99 Equalizer Band 6 Coefficient B2 */ + { 0x9A, 0x00 }, /* 9A Equalizer Band 6 Coefficient A1 */ + { 0x9B, 0x00 }, /* 9B Equalizer Band 6 Coefficient A1 */ + { 0x9C, 0x00 }, /* 9C Equalizer Band 6 Coefficient A1 */ + { 0x9D, 0x00 }, /* 9D Equalizer Band 6 Coefficient A2 */ + { 0x9E, 0x00 }, /* 9E Equalizer Band 6 Coefficient A2 */ + { 0x9F, 0x00 }, /* 9F Equalizer Band 6 Coefficient A2 */ + + { 0xA0, 0x00 }, /* A0 Equalizer Band 7 Coefficient B0 */ + { 0xA1, 0x00 }, /* A1 Equalizer Band 7 Coefficient B0 */ + { 0xA2, 0x00 }, /* A2 Equalizer Band 7 Coefficient B0 */ + { 0xA3, 0x00 }, /* A3 Equalizer Band 7 Coefficient B1 */ + { 0xA4, 0x00 }, /* A4 Equalizer Band 7 Coefficient B1 */ + { 0xA5, 0x00 }, /* A5 Equalizer Band 7 Coefficient B1 */ + { 0xA6, 0x00 }, /* A6 Equalizer Band 7 Coefficient B2 */ + { 0xA7, 0x00 }, /* A7 Equalizer Band 7 Coefficient B2 */ + { 0xA8, 0x00 }, /* A8 Equalizer Band 7 Coefficient B2 */ + { 0xA9, 0x00 }, /* A9 Equalizer Band 7 Coefficient A1 */ + { 0xAA, 0x00 }, /* AA Equalizer Band 7 Coefficient A1 */ + { 0xAB, 0x00 }, /* AB Equalizer Band 7 Coefficient A1 */ + { 0xAC, 0x00 }, /* AC Equalizer Band 7 Coefficient A2 */ + { 0xAD, 0x00 }, /* AD Equalizer Band 7 Coefficient A2 */ + { 0xAE, 0x00 }, /* AE Equalizer Band 7 Coefficient A2 */ + { 0xAF, 0x00 }, /* AF ADC Biquad Coefficient B0 */ + + { 0xB0, 0x00 }, /* B0 ADC Biquad Coefficient B0 */ + { 0xB1, 0x00 }, /* B1 ADC Biquad Coefficient B0 */ + { 0xB2, 0x00 }, /* B2 ADC Biquad Coefficient B1 */ + { 0xB3, 0x00 }, /* B3 ADC Biquad Coefficient B1 */ + { 0xB4, 0x00 }, /* B4 ADC Biquad Coefficient B1 */ + { 0xB5, 0x00 }, /* B5 ADC Biquad Coefficient B2 */ + { 0xB6, 0x00 }, /* B6 ADC Biquad Coefficient B2 */ + { 0xB7, 0x00 }, /* B7 ADC Biquad Coefficient B2 */ + { 0xB8, 0x00 }, /* B8 ADC Biquad Coefficient A1 */ + { 0xB9, 0x00 }, /* B9 ADC Biquad Coefficient A1 */ + { 0xBA, 0x00 }, /* BA ADC Biquad Coefficient A1 */ + { 0xBB, 0x00 }, /* BB ADC Biquad Coefficient A2 */ + { 0xBC, 0x00 }, /* BC ADC Biquad Coefficient A2 */ + { 0xBD, 0x00 }, /* BD ADC Biquad Coefficient A2 */ + { 0xBE, 0x00 }, /* BE Digital Mic 3 Volume */ + { 0xBF, 0x00 }, /* BF Digital Mic 4 Volume */ + + { 0xC0, 0x00 }, /* C0 Digital Mic 34 Biquad Pre Atten */ + { 0xC1, 0x00 }, /* C1 Record TDM Slot */ + { 0xC2, 0x00 }, /* C2 Sample Rate */ + { 0xC3, 0x00 }, /* C3 Digital Mic 34 Biquad Coefficient C3 */ + { 0xC4, 0x00 }, /* C4 Digital Mic 34 Biquad Coefficient C4 */ + { 0xC5, 0x00 }, /* C5 Digital Mic 34 Biquad Coefficient C5 */ + { 0xC6, 0x00 }, /* C6 Digital Mic 34 Biquad Coefficient C6 */ + { 0xC7, 0x00 }, /* C7 Digital Mic 34 Biquad Coefficient C7 */ + { 0xC8, 0x00 }, /* C8 Digital Mic 34 Biquad Coefficient C8 */ + { 0xC9, 0x00 }, /* C9 Digital Mic 34 Biquad Coefficient C9 */ + { 0xCA, 0x00 }, /* CA Digital Mic 34 Biquad Coefficient CA */ + { 0xCB, 0x00 }, /* CB Digital Mic 34 Biquad Coefficient CB */ + { 0xCC, 0x00 }, /* CC Digital Mic 34 Biquad Coefficient CC */ + { 0xCD, 0x00 }, /* CD Digital Mic 34 Biquad Coefficient CD */ + { 0xCE, 0x00 }, /* CE Digital Mic 34 Biquad Coefficient CE */ + { 0xCF, 0x00 }, /* CF Digital Mic 34 Biquad Coefficient CF */ + + { 0xD0, 0x00 }, /* D0 Digital Mic 34 Biquad Coefficient D0 */ + { 0xD1, 0x00 }, /* D1 Digital Mic 34 Biquad Coefficient D1 */ +}; + +static bool max98090_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_REVISION_ID: + return true; + default: + return false; + } +} + +static bool max98090_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_INTERRUPT_S: + case M98090_REG_RESERVED: + case M98090_REG_LINE_INPUT_CONFIG: + case M98090_REG_LINE_INPUT_LEVEL: + case M98090_REG_INPUT_MODE: + case M98090_REG_MIC1_INPUT_LEVEL: + case M98090_REG_MIC2_INPUT_LEVEL: + case M98090_REG_MIC_BIAS_VOLTAGE: + case M98090_REG_DIGITAL_MIC_ENABLE: + case M98090_REG_DIGITAL_MIC_CONFIG: + case M98090_REG_LEFT_ADC_MIXER: + case M98090_REG_RIGHT_ADC_MIXER: + case M98090_REG_LEFT_ADC_LEVEL: + case M98090_REG_RIGHT_ADC_LEVEL: + case M98090_REG_ADC_BIQUAD_LEVEL: + case M98090_REG_ADC_SIDETONE: + case M98090_REG_SYSTEM_CLOCK: + case M98090_REG_CLOCK_MODE: + case M98090_REG_CLOCK_RATIO_NI_MSB: + case M98090_REG_CLOCK_RATIO_NI_LSB: + case M98090_REG_CLOCK_RATIO_MI_MSB: + case M98090_REG_CLOCK_RATIO_MI_LSB: + case M98090_REG_MASTER_MODE: + case M98090_REG_INTERFACE_FORMAT: + case M98090_REG_TDM_CONTROL: + case M98090_REG_TDM_FORMAT: + case M98090_REG_IO_CONFIGURATION: + case M98090_REG_FILTER_CONFIG: + case M98090_REG_DAI_PLAYBACK_LEVEL: + case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: + case M98090_REG_LEFT_HP_MIXER: + case M98090_REG_RIGHT_HP_MIXER: + case M98090_REG_HP_CONTROL: + case M98090_REG_LEFT_HP_VOLUME: + case M98090_REG_RIGHT_HP_VOLUME: + case M98090_REG_LEFT_SPK_MIXER: + case M98090_REG_RIGHT_SPK_MIXER: + case M98090_REG_SPK_CONTROL: + case M98090_REG_LEFT_SPK_VOLUME: + case M98090_REG_RIGHT_SPK_VOLUME: + case M98090_REG_DRC_TIMING: + case M98090_REG_DRC_COMPRESSOR: + case M98090_REG_DRC_EXPANDER: + case M98090_REG_DRC_GAIN: + case M98090_REG_RCV_LOUTL_MIXER: + case M98090_REG_RCV_LOUTL_CONTROL: + case M98090_REG_RCV_LOUTL_VOLUME: + case M98090_REG_LOUTR_MIXER: + case M98090_REG_LOUTR_CONTROL: + case M98090_REG_LOUTR_VOLUME: + case M98090_REG_JACK_DETECT: + case M98090_REG_INPUT_ENABLE: + case M98090_REG_OUTPUT_ENABLE: + case M98090_REG_LEVEL_CONTROL: + case M98090_REG_DSP_FILTER_ENABLE: + case M98090_REG_BIAS_CONTROL: + case M98090_REG_DAC_CONTROL: + case M98090_REG_ADC_CONTROL: + case M98090_REG_DEVICE_SHUTDOWN: + case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: + case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: + case M98090_REG_DMIC3_VOLUME: + case M98090_REG_DMIC4_VOLUME: + case M98090_REG_DMIC34_BQ_PREATTEN: + case M98090_REG_RECORD_TDM_SLOT: + case M98090_REG_SAMPLE_RATE: + case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + return true; + default: + return false; + } +} + +static int max98090_reset(struct max98090_priv *max98090) +{ + int ret; + + /* Reset the codec by writing to this write-only reset register */ + ret = regmap_write(max98090->regmap, M98090_REG_SOFTWARE_RESET, + M98090_SWRESET_MASK); + if (ret < 0) { + dev_err(max98090->codec->dev, + "Failed to reset codec: %d\n", ret); + return ret; + } + + msleep(20); + return ret; +} + +static const unsigned int max98090_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_mic_tlv, 0, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_line_single_ended_tlv, + -600, 600, 0); + +static const unsigned int max98090_line_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-600, 300, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_avg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_av_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_dv_tlv, -1500, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_sidetone_tlv, -6050, 200, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_alc_tlv, -1500, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcmakeup_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alccomp_tlv, -3100, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_drcexp_tlv, -6600, 100, 0); + +static const unsigned int max98090_mixout_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-1200, 250, 0), + 2, 3, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +static const unsigned int max98090_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98090_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 4, TLV_DB_SCALE_ITEM(-4800, 400, 0), + 5, 10, TLV_DB_SCALE_ITEM(-2900, 300, 0), + 11, 14, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 15, 29, TLV_DB_SCALE_ITEM(-500, 100, 0), + 30, 39, TLV_DB_SCALE_ITEM(950, 50, 0), +}; + +static const unsigned int max98090_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + if (val >= 1) { + /* If on, return the volume */ + val = val - 1; + *select = val; + } else { + /* If off, return last stored value */ + val = *select; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + *select = sel; + + /* Setting a volume is only valid if it is already On */ + if (val >= 1) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, mc->reg, + mask << mc->shift, + sel << mc->shift); + + return 0; +} + +static const char * max98090_perf_pwr_text[] = + { "High Performance", "Low Power" }; +static const char * max98090_pwr_perf_text[] = + { "Low Power", "High Performance" }; + +static const struct soc_enum max98090_vcmbandgap_enum = + SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; + +static const struct soc_enum max98090_osr128_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, + ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); + +static const char *max98090_mode_text[] = { "Voice", "Music" }; + +static const struct soc_enum max98090_mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const struct soc_enum max98090_filter_dmic34mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const char * max98090_drcatk_text[] = + { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; + +static const struct soc_enum max98090_drcatk_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, + ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); + +static const char * max98090_drcrls_text[] = + { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; + +static const struct soc_enum max98090_drcrls_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, + ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); + +static const char * max98090_alccmp_text[] = + { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; + +static const struct soc_enum max98090_alccmp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, + ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); + +static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; + +static const struct soc_enum max98090_drcexp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, + ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text); + +static const struct soc_enum max98090_dac_perfmode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT, + ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); + +static const struct soc_enum max98090_dachp_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_adchp_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct snd_kcontrol_new max98090_snd_controls[] = { + SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), + + SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + M98090_MIC_PA1EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + M98090_MIC_PA2EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_TLV("MIC1 Volume", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PGAM1_SHIFT, M98090_MIC_PGAM1_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PGAM2_SHIFT, M98090_MIC_PGAM2_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG135_SHIFT, 0, + M98090_MIXG135_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG246_SHIFT, 0, + M98090_MIXG246_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINAPGA_SHIFT, 0, M98090_LINAPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINBPGA_SHIFT, 0, M98090_LINBPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE("LINEA Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFA_SHIFT, M98090_EXTBUFA_NUM - 1, 0), + SOC_SINGLE("LINEB Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFB_SHIFT, M98090_EXTBUFB_NUM - 1, 0), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVLG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVRG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVL_SHIFT, M98090_AVL_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVR_SHIFT, M98090_AVR_NUM - 1, 1, + max98090_av_tlv), + + SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), + SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CONTROL, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), + SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), + + SOC_SINGLE("DAC Mono Mode", M98090_REG_IO_CONFIGURATION, + M98090_DMONO_SHIFT, M98090_DMONO_NUM - 1, 0), + SOC_SINGLE("SDIN Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, M98090_SDIEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, M98090_SDOEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Hi-Z Mode", M98090_REG_IO_CONFIGURATION, + M98090_HIZOFF_SHIFT, M98090_HIZOFF_NUM - 1, 1), + SOC_ENUM("Filter Mode", max98090_mode_enum), + SOC_SINGLE("Record Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, M98090_AHPF_NUM - 1, 0), + SOC_SINGLE("Playback Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_DHPF_SHIFT, M98090_DHPF_NUM - 1, 0), + SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_ADC_BIQUAD_LEVEL, + M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), + SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", + M98090_REG_ADC_SIDETONE, M98090_DVST_SHIFT, + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + SOC_SINGLE_TLV("Digital Coarse Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVG_SHIFT, M98090_DVG_NUM - 1, 0, + max98090_dvg_tlv), + SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DV_SHIFT, M98090_DV_NUM - 1, 1, + max98090_dv_tlv), + SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQUALIZER_BASE, 105), + SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ Clipping Detection", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_EQCLPN_SHIFT, M98090_EQCLPN_NUM - 1, + 1), + SOC_SINGLE_TLV("Digital EQ Volume", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_DVEQ_SHIFT, M98090_DVEQ_NUM - 1, 1, + max98090_dv_tlv), + + SOC_SINGLE("ALC Enable", M98090_REG_DRC_TIMING, + M98090_DRCEN_SHIFT, M98090_DRCEN_NUM - 1, 0), + SOC_ENUM("ALC Attack Time", max98090_drcatk_enum), + SOC_ENUM("ALC Release Time", max98090_drcrls_enum), + SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_DRC_GAIN, + M98090_DRCG_SHIFT, M98090_DRCG_NUM - 1, 0, + max98090_alcmakeup_tlv), + SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), + SOC_ENUM("ALC Expansion Ratio", max98090_drcexp_enum), + SOC_SINGLE_TLV("ALC Compression Threshold Volume", + M98090_REG_DRC_COMPRESSOR, M98090_DRCTHC_SHIFT, + M98090_DRCTHC_NUM - 1, 1, max98090_alccomp_tlv), + SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_REG_DRC_EXPANDER, M98090_DRCTHE_SHIFT, + M98090_DRCTHE_NUM - 1, 1, max98090_drcexp_tlv), + + SOC_ENUM("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum), + SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + + SOC_SINGLE_TLV("Headphone Left Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPLG_SHIFT, + M98090_MIXHPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Headphone Right Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPRG_SHIFT, + M98090_MIXHPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Speaker Left Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPLG_SHIFT, + M98090_MIXSPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Speaker Right Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPRG_SHIFT, + M98090_MIXSPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Receiver Left Mixer Volume", + M98090_REG_RCV_LOUTL_CONTROL, M98090_MIXRCVLG_SHIFT, + M98090_MIXRCVLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Receiver Right Mixer Volume", + M98090_REG_LOUTR_CONTROL, M98090_MIXRCVRG_SHIFT, + M98090_MIXRCVRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_DOUBLE_R_TLV("Headphone Volume", M98090_REG_LEFT_HP_VOLUME, + M98090_REG_RIGHT_HP_VOLUME, M98090_HPVOLL_SHIFT, + M98090_HPVOLL_NUM - 1, 0, max98090_hp_tlv), + + SOC_DOUBLE_R_RANGE_TLV("Speaker Volume", + M98090_REG_LEFT_SPK_VOLUME, M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPVOLL_SHIFT, 24, M98090_SPVOLL_NUM - 1 + 24, + 0, max98090_spk_tlv), + + SOC_DOUBLE_R_TLV("Receiver Volume", M98090_REG_RCV_LOUTL_VOLUME, + M98090_REG_LOUTR_VOLUME, M98090_RCVLVOL_SHIFT, + M98090_RCVLVOL_NUM - 1, 0, max98090_rcv_lout_tlv), + + SOC_SINGLE("Headphone Left Switch", M98090_REG_LEFT_HP_VOLUME, + M98090_HPLM_SHIFT, 1, 1), + SOC_SINGLE("Headphone Right Switch", M98090_REG_RIGHT_HP_VOLUME, + M98090_HPRM_SHIFT, 1, 1), + + SOC_SINGLE("Speaker Left Switch", M98090_REG_LEFT_SPK_VOLUME, + M98090_SPLM_SHIFT, 1, 1), + SOC_SINGLE("Speaker Right Switch", M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPRM_SHIFT, 1, 1), + + SOC_SINGLE("Receiver Left Switch", M98090_REG_RCV_LOUTL_VOLUME, + M98090_RCVLM_SHIFT, 1, 1), + SOC_SINGLE("Receiver Right Switch", M98090_REG_LOUTR_VOLUME, + M98090_RCVRM_SHIFT, 1, 1), + + SOC_SINGLE("Zero-Crossing Detection", M98090_REG_LEVEL_CONTROL, + M98090_ZDENN_SHIFT, M98090_ZDENN_NUM - 1, 1), + SOC_SINGLE("Enhanced Vol Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VS2ENN_SHIFT, M98090_VS2ENN_NUM - 1, 1), + SOC_SINGLE("Volume Adjustment Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VSENN_SHIFT, M98090_VSENN_NUM - 1, 1), + + SND_SOC_BYTES("Biquad Coefficients", M98090_REG_RECORD_BIQUAD_BASE, 15), + SOC_SINGLE("Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), +}; + +static const struct snd_kcontrol_new max98091_snd_controls[] = { + + SOC_SINGLE("DMIC34 Zeropad", M98090_REG_SAMPLE_RATE, + M98090_DMIC34_ZEROPAD_SHIFT, + M98090_DMIC34_ZEROPAD_NUM - 1, 0), + + SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), + SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34HPF_SHIFT, + M98090_FLT_DMIC34HPF_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("DMIC4 Boost Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4G_SHIFT, M98090_DMIC_AV4G_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("DMIC3 Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3_SHIFT, M98090_DMIC_AV3_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4_SHIFT, M98090_DMIC_AV4_NUM - 1, 1, + max98090_av_tlv), + + SND_SOC_BYTES("DMIC34 Biquad Coefficients", + M98090_REG_DMIC34_BIQUAD_BASE, 15), + SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", + M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, + M98090_AV34BQ_NUM - 1, 1, max98090_dv_tlv), +}; + +static int max98090_micinput_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, w->reg); + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = (val & M98090_MIC_PA1EN_MASK) >> M98090_MIC_PA1EN_SHIFT; + else + val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; + + + if (val >= 1) { + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { + max98090->pa1en = val - 1; /* Update for volatile */ + } else { + max98090->pa2en = val - 1; /* Update for volatile */ + } + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to most recently selected volume */ + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = max98090->pa1en + 1; + else + val = max98090->pa2en + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = 0; + break; + default: + return -EINVAL; + } + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA1EN_MASK, + val << M98090_MIC_PA1EN_SHIFT); + else + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA2EN_MASK, + val << M98090_MIC_PA2EN_SHIFT); + + return 0; +} + +static const char *mic1_mux_text[] = { "IN12", "IN56" }; + +static const struct soc_enum mic1_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT, + ARRAY_SIZE(mic1_mux_text), mic1_mux_text); + +static const struct snd_kcontrol_new max98090_mic1_mux = + SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); + +static const char *mic2_mux_text[] = { "IN34", "IN56" }; + +static const struct soc_enum mic2_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT, + ARRAY_SIZE(mic2_mux_text), mic2_mux_text); + +static const struct snd_kcontrol_new max98090_mic2_mux = + SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); + +static const char * max98090_micpre_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_pa1en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const struct soc_enum max98090_pa2en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +/* LINEA mixer switch */ +static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN1SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN3 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN3SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN5 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN5SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN34DIFF_SHIFT, 1, 0), +}; + +/* LINEB mixer switch */ +static const struct snd_kcontrol_new max98090_lineb_mixer_controls[] = { + SOC_DAPM_SINGLE("IN2 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN2SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN4 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN4SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN6 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN6SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN56DIFF_SHIFT, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98090_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC2_SHIFT, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC2_SHIFT, 1, 0), +}; + +static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; + +static const struct soc_enum ltenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct soc_enum ltenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct snd_kcontrol_new max98090_ltenl_mux = + SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + +static const struct snd_kcontrol_new max98090_ltenr_mux = + SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + +static const char *lben_mux_text[] = { "Normal", "Loopback" }; + +static const struct soc_enum lbenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct soc_enum lbenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct snd_kcontrol_new max98090_lbenl_mux = + SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + +static const struct snd_kcontrol_new max98090_lbenr_mux = + SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + +static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; + +static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" }; + +static const struct soc_enum stenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT, + ARRAY_SIZE(stenl_mux_text), stenl_mux_text); + +static const struct soc_enum stenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT, + ARRAY_SIZE(stenr_mux_text), stenr_mux_text); + +static const struct snd_kcontrol_new max98090_stenl_mux = + SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum); + +static const struct snd_kcontrol_new max98090_stenr_mux = + SOC_DAPM_ENUM("STENR Mux", stenr_mux_enum); + +/* Left speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC2_SHIFT, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC2_SHIFT, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC2_SHIFT, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC2_SHIFT, 1, 0), +}; + +/* Left receiver mixer switch */ +static const struct snd_kcontrol_new max98090_left_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC2_SHIFT, 1, 0), +}; + +/* Right receiver mixer switch */ +static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC2_SHIFT, 1, 0), +}; + +static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; + +static const struct soc_enum linmod_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT, + ARRAY_SIZE(linmod_mux_text), linmod_mux_text); + +static const struct snd_kcontrol_new max98090_linmod_mux = + SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); + +static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; + +/* + * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable + */ +static const struct soc_enum mixhplsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhplsel_mux = + SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); + +static const struct soc_enum mixhprsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhprsel_mux = + SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); + +static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMICL"), + SND_SOC_DAPM_INPUT("DMICR"), + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN12"), + SND_SOC_DAPM_INPUT("IN34"), + SND_SOC_DAPM_INPUT("IN56"), + + SND_SOC_DAPM_SUPPLY("MICBIAS", M98090_REG_INPUT_ENABLE, + M98090_MBEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICR_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, 0, NULL, 0), + +/* + * Note: Sysclk and misc power supplies are taken care of by SHDN + */ + + SND_SOC_DAPM_MUX("MIC1 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic1_mux), + + SND_SOC_DAPM_MUX("MIC2 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic2_mux), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PA2EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("LINEA Mixer", SND_SOC_NOPM, 0, 0, + &max98090_linea_mixer_controls[0], + ARRAY_SIZE(max98090_linea_mixer_controls)), + + SND_SOC_DAPM_MIXER("LINEB Mixer", SND_SOC_NOPM, 0, 0, + &max98090_lineb_mixer_controls[0], + ARRAY_SIZE(max98090_lineb_mixer_controls)), + + SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_INPUT_ENABLE, + M98090_LINEAEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_INPUT_ENABLE, + M98090_LINEBEN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_adc_mixer_controls[0], + ARRAY_SIZE(max98090_left_adc_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_adc_mixer_controls[0], + ARRAY_SIZE(max98090_right_adc_mixer_controls)), + + SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "HiFi Capture", 1, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("LBENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenl_mux), + + SND_SOC_DAPM_MUX("LBENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenr_mux), + + SND_SOC_DAPM_MUX("LTENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenl_mux), + + SND_SOC_DAPM_MUX("LTENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenr_mux), + + SND_SOC_DAPM_MUX("STENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenl_mux), + + SND_SOC_DAPM_MUX("STENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenr_mux), + + SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DALEN_SHIFT, 0), + SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DAREN_SHIFT, 0), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_hp_mixer_controls[0], + ARRAY_SIZE(max98090_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_hp_mixer_controls[0], + ARRAY_SIZE(max98090_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_left_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_right_rcv_mixer_controls)), + + SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, + M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + + SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCVL"), + SND_SOC_DAPM_OUTPUT("RCVR"), +}; + +static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC3_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC4_SHIFT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, + + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, + {"DMICL", NULL, "AHPF"}, + {"DMICR", NULL, "AHPF"}, + + /* MIC1 input mux */ + {"MIC1 Mux", "IN12", "IN12"}, + {"MIC1 Mux", "IN56", "IN56"}, + + /* MIC2 input mux */ + {"MIC2 Mux", "IN34", "IN34"}, + {"MIC2 Mux", "IN56", "IN56"}, + + {"MIC1 Input", NULL, "MIC1 Mux"}, + {"MIC2 Input", NULL, "MIC2 Mux"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "IN12 Switch", "IN12"}, + {"Left ADC Mixer", "IN34 Switch", "IN34"}, + {"Left ADC Mixer", "IN56 Switch", "IN56"}, + {"Left ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Left ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "IN12 Switch", "IN12"}, + {"Right ADC Mixer", "IN34 Switch", "IN34"}, + {"Right ADC Mixer", "IN56 Switch", "IN56"}, + {"Right ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Right ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Line A input mixer */ + {"LINEA Mixer", "IN1 Switch", "IN1"}, + {"LINEA Mixer", "IN3 Switch", "IN3"}, + {"LINEA Mixer", "IN5 Switch", "IN5"}, + {"LINEA Mixer", "IN34 Switch", "IN34"}, + + /* Line B input mixer */ + {"LINEB Mixer", "IN2 Switch", "IN2"}, + {"LINEB Mixer", "IN4 Switch", "IN4"}, + {"LINEB Mixer", "IN6 Switch", "IN6"}, + {"LINEB Mixer", "IN56 Switch", "IN56"}, + + {"LINEA Input", NULL, "LINEA Mixer"}, + {"LINEB Input", NULL, "LINEB Mixer"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"ADCL", NULL, "SHDN"}, + {"ADCR", NULL, "SHDN"}, + + {"LBENL Mux", "Normal", "ADCL"}, + {"LBENL Mux", "Normal", "DMICL"}, + {"LBENL Mux", "Loopback", "LTENL Mux"}, + {"LBENR Mux", "Normal", "ADCR"}, + {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Loopback", "LTENR Mux"}, + + {"AIFOUTL", NULL, "LBENL Mux"}, + {"AIFOUTR", NULL, "LBENR Mux"}, + {"AIFOUTL", NULL, "SHDN"}, + {"AIFOUTR", NULL, "SHDN"}, + {"AIFOUTL", NULL, "SDOEN"}, + {"AIFOUTR", NULL, "SDOEN"}, + + {"LTENL Mux", "Normal", "AIFINL"}, + {"LTENL Mux", "Loopthrough", "LBENL Mux"}, + {"LTENR Mux", "Normal", "AIFINR"}, + {"LTENR Mux", "Loopthrough", "LBENR Mux"}, + + {"DACL", NULL, "LTENL Mux"}, + {"DACR", NULL, "LTENR Mux"}, + + {"STENL Mux", "Sidetone Left", "ADCL"}, + {"STENL Mux", "Sidetone Left", "DMICL"}, + {"STENR Mux", "Sidetone Right", "ADCR"}, + {"STENR Mux", "Sidetone Right", "DMICR"}, + {"DACL", "NULL", "STENL Mux"}, + {"DACR", "NULL", "STENL Mux"}, + + {"AIFINL", NULL, "SHDN"}, + {"AIFINR", NULL, "SHDN"}, + {"AIFINL", NULL, "SDIEN"}, + {"AIFINR", NULL, "SDIEN"}, + {"DACL", NULL, "SHDN"}, + {"DACR", NULL, "SHDN"}, + + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Left Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Right Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Left Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Right Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left Receiver output mixer */ + {"Left Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Left Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Left Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right Receiver output mixer */ + {"Right Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Right Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Right Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + {"MIXHPLSEL Mux", "HP Mixer", "Left Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Left Out", NULL, "DACL"}, + {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + + {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Right Out", NULL, "DACR"}, + {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Left Out", NULL, "Left Receiver Mixer"}, + + {"LINMOD Mux", "Left and Right", "Right Receiver Mixer"}, + {"LINMOD Mux", "Left Only", "Left Receiver Mixer"}, + {"RCV Right Out", NULL, "LINMOD Mux"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCVL", NULL, "RCV Left Out"}, + {"RCVR", NULL, "RCV Right Out"}, + +}; + +static const struct snd_soc_dapm_route max98091_dapm_routes[] = { + + /* DMIC inputs */ + {"DMIC3", NULL, "DMIC3_ENA"}, + {"DMIC4", NULL, "DMIC4_ENA"}, + {"DMIC3", NULL, "AHPF"}, + {"DMIC4", NULL, "AHPF"}, + +}; + +static int max98090_add_widgets(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_add_codec_controls(codec, max98090_snd_controls, + ARRAY_SIZE(max98090_snd_controls)); + + if (max98090->devtype == MAX98091) { + snd_soc_add_codec_controls(codec, max98091_snd_controls, + ARRAY_SIZE(max98091_snd_controls)); + } + + snd_soc_dapm_new_controls(dapm, max98090_dapm_widgets, + ARRAY_SIZE(max98090_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98090_dapm_routes, + ARRAY_SIZE(max98090_dapm_routes)); + + if (max98090->devtype == MAX98091) { + snd_soc_dapm_new_controls(dapm, max98091_dapm_widgets, + ARRAY_SIZE(max98091_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, + ARRAY_SIZE(max98091_dapm_routes)); + + } + + return 0; +} + +static const int pclk_rates[] = { + 12000000, 12000000, 13000000, 13000000, + 16000000, 16000000, 19200000, 19200000 +}; + +static const int lrclk_rates[] = { + 8000, 16000, 8000, 16000, + 8000, 16000, 8000, 16000 +}; + +static const int user_pclk_rates[] = { + 13000000, 13000000 +}; + +static const int user_lrclk_rates[] = { + 44100, 48000 +}; + +static const unsigned long long ni_value[] = { + 3528, 768 +}; + +static const unsigned long long mi_value[] = { + 8125, 1625 +}; + +static void max98090_configure_bclk(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned long long ni; + int i; + + if (!max98090->sysclk) { + dev_err(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!max98090->bclk || !max98090->lrclk) { + dev_err(codec->dev, "No audio clocks configured\n"); + return; + } + + /* Skip configuration when operating as slave */ + if (!(snd_soc_read(codec, M98090_REG_MASTER_MODE) & + M98090_MAS_MASK)) { + return; + } + + /* Check for supported PCLK to LRCLK ratios */ + for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { + if ((pclk_rates[i] == max98090->sysclk) && + (lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found supported PCLK to LRCLK rates 0x%x\n", + i + 0x8); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, + (i + 0x8) << M98090_FREQ_SHIFT); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + return; + } + } + + /* Check for user calculated MI and NI ratios */ + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { + if ((user_pclk_rates[i] == max98090->sysclk) && + (user_lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found user supported PCLK to LRCLK rates\n"); + dev_dbg(codec->dev, "i %d ni %lld mi %lld\n", + i, ni_value[i], mi_value[i]); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, + 1 << M98090_USE_M1_SHIFT); + + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, + ni_value[i] & 0xFF); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_MSB, + (mi_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_LSB, + mi_value[i] & 0xFF); + + return; + } + } + + /* + * Calculate based on MI = 65536 (not as good as either method above) + */ + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + + /* + * Configure NI when operating as master + * Note: There is a small, but significant audio quality improvement + * by calculating ni and mi. + */ + ni = 65536ULL * (max98090->lrclk < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)max98090->lrclk; + do_div(ni, (unsigned long long int)max98090->sysclk); + dev_info(codec->dev, "No better method found\n"); + dev_info(codec->dev, "Calculating ni %lld with mi 65536\n", ni); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, ni & 0xFF); +} + +static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + u8 regval; + + max98090->dai_fmt = fmt; + cdata = &max98090->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Set to slave mode PLL - MAS mode off */ + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_MSB, 0x00); + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_LSB, 0x00); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + if (max98090->tdm_slots == 4) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_64; + } else if (max98090->tdm_slots == 3) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_48; + } else { + /* Few TDM slots, or No TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_32; + } + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + snd_soc_write(codec, M98090_REG_MASTER_MODE, regval); + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98090_DLY_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regval |= M98090_RJ_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Not supported mode */ + default: + dev_err(codec->dev, "DAI format unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98090_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98090_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98090_BCI_MASK|M98090_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + /* + * This accommodates an inverted logic in the MAX98090 chip + * for Bit Clock Invert (BCI). The inverted logic is only + * seen for the case of TDM mode. The remaining cases have + * normal logic. + */ + if (max98090->tdm_slots > 1) { + regval ^= M98090_BCI_MASK; + } + + snd_soc_write(codec, + M98090_REG_INTERFACE_FORMAT, regval); + } + + return 0; +} + +static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; + + if (slots < 0 || slots > 4) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_width = slot_width; + + if (max98090->tdm_slots > 1) { + /* SLOTL SLOTR SLOTDLY */ + snd_soc_write(codec, M98090_REG_TDM_FORMAT, + 0 << M98090_TDM_SLOTL_SHIFT | + 1 << M98090_TDM_SLOTR_SHIFT | + 0 << M98090_TDM_SLOTDLY_SHIFT); + + /* FSW TDM */ + snd_soc_update_bits(codec, M98090_REG_TDM_CONTROL, + M98090_TDM_MASK, + M98090_TDM_MASK); + } + + /* + * Normally advisable to set TDM first, but this permits either order + */ + cdata->fmt = 0; + max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + + return 0; +} + +static int max98090_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { + /* + * Set to normal bias level. + */ + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, + M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + } + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + /* Set internal pull-up to lowest power mode */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + regcache_mark_dirty(max98090->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const int comp_pclk_rates[] = { + 11289600, 12288000, 12000000, 13000000, 19200000 +}; + +static const int dmic_micclk[] = { + 2, 2, 2, 2, 4, 2 +}; + +static const int comp_lrclk_rates[] = { + 8000, 16000, 32000, 44100, 48000, 96000 +}; + +static const int dmic_comp[6][6] = { + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 1, 1, 1}, + {7, 8, 3, 1, 2, 2}, + {7, 8, 3, 3, 3, 3} +}; + +static int max98090_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int i, j; + + cdata = &max98090->dai[0]; + max98090->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + max98090->bclk *= 2; + + max98090->lrclk = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98090_REG_INTERFACE_FORMAT, + M98090_WS_MASK, 0); + break; + default: + return -EINVAL; + } + + max98090_configure_bclk(codec); + + cdata->rate = max98090->lrclk; + + /* Update filter mode */ + if (max98090->lrclk < 24000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, M98090_MODE_MASK); + + /* Update sample rate mode */ + if (max98090->lrclk < 50000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, M98090_DHF_MASK); + + /* Check for supported PCLK to LRCLK ratios */ + for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { + if (comp_pclk_rates[j] == max98090->sysclk) { + break; + } + } + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (max98090->lrclk <= (comp_lrclk_rates[i] + + comp_lrclk_rates[i + 1]) / 2) { + break; + } + } + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE, + M98090_MICCLK_MASK, + dmic_micclk[j] << M98090_MICCLK_SHIFT); + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_MASK, + dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + + return 0; +} + +/* + * PLL / Sysclk + */ +static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98090->sysclk) + return 0; + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV1); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV2); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV4); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + max98090->sysclk = freq; + + max98090_configure_bclk(codec); + + return 0; +} + +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int regval; + + regval = mute ? M98090_DVM_MASK : 0; + snd_soc_update_bits(codec, M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVM_MASK, regval); + + return 0; +} + +static void max98090_jack_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = container_of(work, + struct max98090_priv, + jack_work.work); + struct snd_soc_codec *codec = max98090->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int status = 0; + int reg; + + /* Read a second time */ + if (max98090->jack_state == M98090_JACK_STATE_NO_HEADSET) { + + /* Strong pull up allows mic detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, 0); + + msleep(50); + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + /* Weak pull up allows only insertion detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + } else { + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + } + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { + case M98090_LSNS_MASK | M98090_JKSNS_MASK: + dev_dbg(codec->dev, "No Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + status |= 0; + + break; + + case 0: + if (max98090->jack_state == + M98090_JACK_STATE_HEADSET) { + + dev_dbg(codec->dev, + "Headset Button Down Detected\n"); + + /* + * max98090_headset_button_event(codec) + * could be defined, then called here. + */ + + status |= SND_JACK_HEADSET; + status |= SND_JACK_BTN_0; + + break; + } + + /* Line is reported as Headphone */ + /* Nokia Headset is reported as Headphone */ + /* Mono Headphone is reported as Headphone */ + dev_dbg(codec->dev, "Headphone Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADPHONE; + + status |= SND_JACK_HEADPHONE; + + break; + + case M98090_JKSNS_MASK: + dev_dbg(codec->dev, "Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADSET; + + status |= SND_JACK_HEADSET; + + break; + + default: + dev_dbg(codec->dev, "Unrecognized Jack Status\n"); + break; + } + + snd_soc_jack_report(max98090->jack, status, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + snd_soc_dapm_sync(dapm); +} + +static irqreturn_t max98090_interrupt(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int mask; + unsigned int active; + + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); + + ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_INTERRUPT_S: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &active); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_DEVICE_STATUS: %d\n", + ret); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "active=0x%02x mask=0x%02x -> active=0x%02x\n", + active, mask, active & mask); + + active &= mask; + + if (!active) + return IRQ_NONE; + + if (active & M98090_CLD_MASK) { + dev_err(codec->dev, "M98090_CLD_MASK\n"); + } + + if (active & M98090_SLD_MASK) { + dev_dbg(codec->dev, "M98090_SLD_MASK\n"); + } + + if (active & M98090_ULK_MASK) { + dev_err(codec->dev, "M98090_ULK_MASK\n"); + } + + if (active & M98090_JDET_MASK) { + dev_dbg(codec->dev, "M98090_JDET_MASK\n"); + + pm_wakeup_event(codec->dev, 100); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + } + + if (active & M98090_DRCACT_MASK) { + dev_dbg(codec->dev, "M98090_DRCACT_MASK\n"); + } + + if (active & M98090_DRCCLP_MASK) { + dev_err(codec->dev, "M98090_DRCCLP_MASK\n"); + } + + return IRQ_HANDLED; +} + +/** + * max98090_mic_detect - Enable microphone detection via the MAX98090 IRQ + * + * @codec: MAX98090 codec + * @jack: jack to report detection events on + * + * Enable microphone detection via IRQ on the MAX98090. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for MAX98090 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * If no jack is supplied detection will be disabled. + */ +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "max98090_mic_detect\n"); + + max98090->jack = jack; + if (jack) { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 1 << M98090_IJDET_SHIFT); + } else { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 0); + } + + /* Send an initial empty report */ + snd_soc_jack_report(max98090->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + + return 0; +} +EXPORT_SYMBOL_GPL(max98090_mic_detect); + +#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98090_dai_ops = { + .set_sysclk = max98090_dai_set_sysclk, + .set_fmt = max98090_dai_set_fmt, + .set_tdm_slot = max98090_set_tdm_slot, + .hw_params = max98090_dai_hw_params, + .digital_mute = max98090_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98090_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .ops = &max98090_dai_ops, +} +}; + +static void max98090_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_pdata *pdata = max98090->pdata; + + if (!pdata) { + dev_err(codec->dev, "No platform data\n"); + return; + } + +} + +static int max98090_probe(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int ret = 0; + + dev_dbg(codec->dev, "max98090_probe\n"); + + max98090->codec = codec; + + codec->control_data = max98090->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Reset the codec, the DSP core, and disable all interrupts */ + max98090_reset(max98090); + + /* Initialize private data */ + + max98090->sysclk = (unsigned)-1; + + cdata = &max98090->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98090->lin_state = 0; + max98090->pa1en = 0; + max98090->pa2en = 0; + max98090->extmic_mux = 0; + + ret = snd_soc_read(codec, M98090_REG_REVISION_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + + if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { + max98090->devtype = MAX98090; + dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); + } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { + max98090->devtype = MAX98091; + dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); + } else { + max98090->devtype = MAX98090; + dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); + } + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + + /* Enable jack detection */ + snd_soc_write(codec, M98090_REG_JACK_DETECT, + M98090_JDETEN_MASK | M98090_JDEB_25MS); + + /* Register for interrupts */ + dev_dbg(codec->dev, "irq = %d\n", max98090->irq); + + ret = request_threaded_irq(max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING, + "max98090_interrupt", codec); + if (ret < 0) { + dev_err(codec->dev, "request_irq failed: %d\n", + ret); + } + + /* + * Clear any old interrupts. + * An old interrupt ocurring prior to installing the ISR + * can keep a new interrupt from generating a trigger. + */ + snd_soc_read(codec, M98090_REG_DEVICE_STATUS); + + /* High Performance is default */ + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_DACHP_MASK, + 1 << M98090_DACHP_SHIFT); + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_PERFMODE_MASK, + 0 << M98090_PERFMODE_SHIFT); + snd_soc_update_bits(codec, M98090_REG_ADC_CONTROL, + M98090_ADCHP_MASK, + 1 << M98090_ADCHP_SHIFT); + + /* Turn on VCM bandgap reference */ + snd_soc_write(codec, M98090_REG_BIAS_CONTROL, + M98090_VCM_MODE_MASK); + + max98090_handle_pdata(codec); + + max98090_add_widgets(codec); + +err_access: + return ret; +} + +static int max98090_remove(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&max98090->jack_work); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98090 = { + .probe = max98090_probe, + .remove = max98090_remove, + .set_bias_level = max98090_set_bias_level, +}; + +static const struct regmap_config max98090_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX98090_MAX_REGISTER, + .reg_defaults = max98090_reg, + .num_reg_defaults = ARRAY_SIZE(max98090_reg), + .volatile_reg = max98090_volatile_register, + .readable_reg = max98090_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98090_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98090_priv *max98090; + int ret; + + pr_debug("max98090_i2c_probe\n"); + + max98090 = devm_kzalloc(&i2c->dev, sizeof(struct max98090_priv), + GFP_KERNEL); + if (max98090 == NULL) + return -ENOMEM; + + max98090->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98090); + max98090->control_data = i2c; + max98090->pdata = i2c->dev.platform_data; + max98090->irq = i2c->irq; + + max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + if (IS_ERR(max98090->regmap)) { + ret = PTR_ERR(max98090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98090, max98090_dai, + ARRAY_SIZE(max98090_dai)); + if (ret < 0) + regmap_exit(max98090->regmap); + +err_enable: + return ret; +} + +static int max98090_i2c_remove(struct i2c_client *client) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(max98090->regmap); + return 0; +} + +static int max98090_runtime_resume(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, false); + + regcache_sync(max98090->regmap); + + return 0; +} + +static int max98090_runtime_suspend(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, true); + + return 0; +} + +static struct dev_pm_ops max98090_pm = { + SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + max98090_runtime_resume, NULL) +}; + +static const struct i2c_device_id max98090_i2c_id[] = { + { "max98090", MAX98090 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); + +static struct i2c_driver max98090_i2c_driver = { + .driver = { + .name = "max98090", + .owner = THIS_MODULE, + .pm = &max98090_pm, + }, + .probe = max98090_i2c_probe, + .remove = max98090_i2c_remove, + .id_table = max98090_i2c_id, +}; + +module_i2c_driver(max98090_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroqin, Jerry Wong"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h new file mode 100755 index 000000000000..7e103f249053 --- /dev/null +++ b/sound/soc/codecs/max98090.h @@ -0,0 +1,1549 @@ +/* + * max98090.h -- MAX98090 ALSA SoC Audio driver + * + * 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 version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98090_H +#define _MAX98090_H + +#include + +/* One can override the Linux version here with an explicit version number */ +#define M98090_LINUX_VERSION LINUX_VERSION_CODE + +/* + * MAX98090 Register Definitions + */ + +#define M98090_REG_SOFTWARE_RESET 0x00 +#define M98090_REG_DEVICE_STATUS 0x01 +#define M98090_REG_JACK_STATUS 0x02 +#define M98090_REG_INTERRUPT_S 0x03 +#define M98090_REG_QUICK_SYSTEM_CLOCK 0x04 +#define M98090_REG_QUICK_SAMPLE_RATE 0x05 +#define M98090_REG_DAI_INTERFACE 0x06 +#define M98090_REG_DAC_PATH 0x07 +#define M98090_REG_MIC_DIRECT_TO_ADC 0x08 +#define M98090_REG_LINE_TO_ADC 0x09 +#define M98090_REG_ANALOG_MIC_LOOP 0x0A +#define M98090_REG_ANALOG_LINE_LOOP 0x0B +#define M98090_REG_RESERVED 0x0C +#define M98090_REG_LINE_INPUT_CONFIG 0x0D +#define M98090_REG_LINE_INPUT_LEVEL 0x0E +#define M98090_REG_INPUT_MODE 0x0F +#define M98090_REG_MIC1_INPUT_LEVEL 0x10 +#define M98090_REG_MIC2_INPUT_LEVEL 0x11 +#define M98090_REG_MIC_BIAS_VOLTAGE 0x12 +#define M98090_REG_DIGITAL_MIC_ENABLE 0x13 +#define M98090_REG_DIGITAL_MIC_CONFIG 0x14 +#define M98090_REG_LEFT_ADC_MIXER 0x15 +#define M98090_REG_RIGHT_ADC_MIXER 0x16 +#define M98090_REG_LEFT_ADC_LEVEL 0x17 +#define M98090_REG_RIGHT_ADC_LEVEL 0x18 +#define M98090_REG_ADC_BIQUAD_LEVEL 0x19 +#define M98090_REG_ADC_SIDETONE 0x1A +#define M98090_REG_SYSTEM_CLOCK 0x1B +#define M98090_REG_CLOCK_MODE 0x1C +#define M98090_REG_CLOCK_RATIO_NI_MSB 0x1D +#define M98090_REG_CLOCK_RATIO_NI_LSB 0x1E +#define M98090_REG_CLOCK_RATIO_MI_MSB 0x1F +#define M98090_REG_CLOCK_RATIO_MI_LSB 0x20 +#define M98090_REG_MASTER_MODE 0x21 +#define M98090_REG_INTERFACE_FORMAT 0x22 +#define M98090_REG_TDM_CONTROL 0x23 +#define M98090_REG_TDM_FORMAT 0x24 +#define M98090_REG_IO_CONFIGURATION 0x25 +#define M98090_REG_FILTER_CONFIG 0x26 +#define M98090_REG_DAI_PLAYBACK_LEVEL 0x27 +#define M98090_REG_DAI_PLAYBACK_LEVEL_EQ 0x28 +#define M98090_REG_LEFT_HP_MIXER 0x29 +#define M98090_REG_RIGHT_HP_MIXER 0x2A +#define M98090_REG_HP_CONTROL 0x2B +#define M98090_REG_LEFT_HP_VOLUME 0x2C +#define M98090_REG_RIGHT_HP_VOLUME 0x2D +#define M98090_REG_LEFT_SPK_MIXER 0x2E +#define M98090_REG_RIGHT_SPK_MIXER 0x2F +#define M98090_REG_SPK_CONTROL 0x30 +#define M98090_REG_LEFT_SPK_VOLUME 0x31 +#define M98090_REG_RIGHT_SPK_VOLUME 0x32 +#define M98090_REG_DRC_TIMING 0x33 +#define M98090_REG_DRC_COMPRESSOR 0x34 +#define M98090_REG_DRC_EXPANDER 0x35 +#define M98090_REG_DRC_GAIN 0x36 +#define M98090_REG_RCV_LOUTL_MIXER 0x37 +#define M98090_REG_RCV_LOUTL_CONTROL 0x38 +#define M98090_REG_RCV_LOUTL_VOLUME 0x39 +#define M98090_REG_LOUTR_MIXER 0x3A +#define M98090_REG_LOUTR_CONTROL 0x3B +#define M98090_REG_LOUTR_VOLUME 0x3C +#define M98090_REG_JACK_DETECT 0x3D +#define M98090_REG_INPUT_ENABLE 0x3E +#define M98090_REG_OUTPUT_ENABLE 0x3F +#define M98090_REG_LEVEL_CONTROL 0x40 +#define M98090_REG_DSP_FILTER_ENABLE 0x41 +#define M98090_REG_BIAS_CONTROL 0x42 +#define M98090_REG_DAC_CONTROL 0x43 +#define M98090_REG_ADC_CONTROL 0x44 +#define M98090_REG_DEVICE_SHUTDOWN 0x45 +#define M98090_REG_EQUALIZER_BASE 0x46 +#define M98090_REG_RECORD_BIQUAD_BASE 0xAF +#define M98090_REG_DMIC3_VOLUME 0xBE +#define M98090_REG_DMIC4_VOLUME 0xBF +#define M98090_REG_DMIC34_BQ_PREATTEN 0xC0 +#define M98090_REG_RECORD_TDM_SLOT 0xC1 +#define M98090_REG_SAMPLE_RATE 0xC2 +#define M98090_REG_DMIC34_BIQUAD_BASE 0xC3 +#define M98090_REG_REVISION_ID 0xFF + +#define M98090_REG_CNT (0xFF+1) +#define MAX98090_MAX_REGISTER 0xFF + +/* MAX98090 Register Bit Fields */ + +/* + * M98090_REG_SOFTWARE_RESET + */ +#define M98090_SWRESET_MASK (1<<7) +#define M98090_SWRESET_SHIFT 7 +#define M98090_SWRESET_WIDTH 1 + +/* + * M98090_REG_DEVICE_STATUS + */ +#define M98090_CLD_MASK (1<<7) +#define M98090_CLD_SHIFT 7 +#define M98090_CLD_WIDTH 1 +#define M98090_SLD_MASK (1<<6) +#define M98090_SLD_SHIFT 6 +#define M98090_SLD_WIDTH 1 +#define M98090_ULK_MASK (1<<5) +#define M98090_ULK_SHIFT 5 +#define M98090_ULK_WIDTH 1 +#define M98090_JDET_MASK (1<<2) +#define M98090_JDET_SHIFT 2 +#define M98090_JDET_WIDTH 1 +#define M98090_DRCACT_MASK (1<<1) +#define M98090_DRCACT_SHIFT 1 +#define M98090_DRCACT_WIDTH 1 +#define M98090_DRCCLP_MASK (1<<0) +#define M98090_DRCCLP_SHIFT 0 +#define M98090_DRCCLP_WIDTH 1 + +/* + * M98090_REG_JACK_STATUS + */ +#define M98090_LSNS_MASK (1<<2) +#define M98090_LSNS_SHIFT 2 +#define M98090_LSNS_WIDTH 1 +#define M98090_JKSNS_MASK (1<<1) +#define M98090_JKSNS_SHIFT 1 +#define M98090_JKSNS_WIDTH 1 + +/* + * M98090_REG_INTERRUPT_S + */ +#define M98090_ICLD_MASK (1<<7) +#define M98090_ICLD_SHIFT 7 +#define M98090_ICLD_WIDTH 1 +#define M98090_ISLD_MASK (1<<6) +#define M98090_ISLD_SHIFT 6 +#define M98090_ISLD_WIDTH 1 +#define M98090_IULK_MASK (1<<5) +#define M98090_IULK_SHIFT 5 +#define M98090_IULK_WIDTH 1 +#define M98090_IJDET_MASK (1<<2) +#define M98090_IJDET_SHIFT 2 +#define M98090_IJDET_WIDTH 1 +#define M98090_IDRCACT_MASK (1<<1) +#define M98090_IDRCACT_SHIFT 1 +#define M98090_IDRCACT_WIDTH 1 +#define M98090_IDRCCLP_MASK (1<<0) +#define M98090_IDRCCLP_SHIFT 0 +#define M98090_IDRCCLP_WIDTH 1 + +/* + * M98090_REG_QUICK_SYSTEM_CLOCK + */ +#define M98090_26M_MASK (1<<7) +#define M98090_26M_SHIFT 7 +#define M98090_26M_WIDTH 1 +#define M98090_19P2M_MASK (1<<6) +#define M98090_19P2M_SHIFT 6 +#define M98090_19P2M_WIDTH 1 +#define M98090_13M_MASK (1<<5) +#define M98090_13M_SHIFT 5 +#define M98090_13M_WIDTH 1 +#define M98090_12P288M_MASK (1<<4) +#define M98090_12P288M_SHIFT 4 +#define M98090_12P288M_WIDTH 1 +#define M98090_12M_MASK (1<<3) +#define M98090_12M_SHIFT 3 +#define M98090_12M_WIDTH 1 +#define M98090_11P2896M_MASK (1<<2) +#define M98090_11P2896M_SHIFT 2 +#define M98090_11P2896M_WIDTH 1 +#define M98090_256FS_MASK (1<<0) +#define M98090_256FS_SHIFT 0 +#define M98090_256FS_WIDTH 1 +#define M98090_CLK_ALL_SHIFT 0 +#define M98090_CLK_ALL_WIDTH 8 +#define M98090_CLK_ALL_NUM (1<> 8) & 0xff) +#define M98090_BYTE0(w) (w & 0xff) + +/* Silicon revision number */ +#define M98090_REVA 0x40 +#define M98091_REVA 0x50 + +enum max98090_type { + MAX98090, + MAX98091, +}; + +struct max98090_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98090_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + enum max98090_type devtype; + void *control_data; + struct max98090_pdata *pdata; + unsigned int sysclk; + unsigned int bclk; + unsigned int lrclk; + struct max98090_cdata dai[1]; + int irq; + int jack_state; + struct delayed_work jack_work; + struct snd_soc_jack *jack; + unsigned int dai_fmt; + int tdm_slots; + int tdm_width; + u8 lin_state; + unsigned int pa1en; + unsigned int pa2en; + unsigned int extmic_mux; + unsigned int sidetone; +}; + +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- cgit v1.2.3 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 +++- sound/soc/soc-compress.c | 19 +++++++++---------- sound/soc/soc-core.c | 12 ++++++++++-- sound/soc/soc-dapm.c | 6 ++++-- sound/soc/soc-pcm.c | 7 +++---- 5 files changed, 29 insertions(+), 19 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3953cea0ecfb..a680f23a04fb 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. diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 35726cbf1f02..b5b3db71e253 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,12 +116,13 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; - snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + cpu_dai->active--; codec_dai->active--; codec->active--; @@ -178,15 +179,13 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - snd_soc_dai_digital_mute(codec_dai, 0); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_soc_dai_digital_mute(codec_dai, 1); - break; - } + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0, cstream->direction); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + break; } out: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2370063b5824..4eac22797893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3540,12 +3540,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); * snd_soc_dai_digital_mute - configure DAI system or master clock. * @dai: DAI * @mute: mute enable + * @direction: stream to mute * * Mutes the DAI DAC. */ -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) { - if (dai->driver && dai->driver->ops->digital_mute) + if (!dai->driver) + return -ENOTSUPP; + + if (dai->driver->ops->mute_stream) + return dai->driver->ops->mute_stream(dai, mute, direction); + else if (direction == SNDRV_PCM_STREAM_PLAYBACK && + dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else return -ENOTSUPP; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1e36bc81e5af..4d664f3df805 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3247,14 +3247,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0); + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); ret = 0; break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1); + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); ret = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6aebbe..d675b4ae0df6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -383,8 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -488,7 +487,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -586,7 +585,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) -- cgit v1.2.3 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 --- Documentation/sound/alsa/compress_offload.txt | 46 +++++++++++++ include/sound/compress_driver.h | 8 +++ include/uapi/sound/compress_offload.h | 31 ++++++++- sound/core/compress_offload.c | 96 +++++++++++++++++++++++++++ 4 files changed, 180 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a11abc..0bcc55155911 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this 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/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a43fedf..d630163b9a2e 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 diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 2d620688cfb7..c84abc886e90 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -486,6 +486,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->metadata_set = false; + stream->next_track = false; } else { return -EPERM; } @@ -517,6 +519,49 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->get_metadata(stream, &metadata); + if (retval != 0) + return retval; + + if (copy_to_user((void __user *)arg, &metadata, sizeof(metadata))) + return -EFAULT; + + return 0; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->set_metadata(stream, &metadata); + stream->metadata_set = true; + + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -600,6 +645,44 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -629,6 +712,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -650,6 +739,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- cgit v1.2.3 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 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7213.c | 1599 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7213.h | 523 +++++++++++++++ 5 files changed, 2180 insertions(+) create mode 100644 include/sound/da7213.h create mode 100644 sound/soc/codecs/da7213.c create mode 100644 sound/soc/codecs/da7213.h (limited to 'include') 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 + * + * 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/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a847828932a..751476aa7814 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DFBMCS320 @@ -247,6 +248,9 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DA7213 + tristate + config SND_SOC_DA732X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36cceb7..6a3b3c3b8b41 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-dfbmcs320-objs := dfbmcs320.o @@ -147,6 +148,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c new file mode 100644 index 000000000000..41230ad1c3e0 --- /dev/null +++ b/sound/soc/codecs/da7213.c @@ -0,0 +1,1599 @@ +/* + * DA7213 ALSA SoC Codec Driver + * + * Copyright (c) 2013 Dialog Semiconductor + * + * Author: Adam Thomson + * Based on DA9055 ALSA SoC codec driver. + * + * 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "da7213.h" + + +/* Gain and Volume */ +static const unsigned int aux_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + /* -54dB */ + 0x0, 0x11, TLV_DB_SCALE_ITEM(-5400, 0, 0), + /* -52.5dB to 15dB */ + 0x12, 0x3f, TLV_DB_SCALE_ITEM(-5250, 150, 0) +}; + +static const unsigned int digital_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -78dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7800, 75, 0) +}; + +static const unsigned int alc_analog_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* 0dB to 36dB */ + 0x01, 0x07, TLV_DB_SCALE_ITEM(0, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(lineout_vol_tlv, -4800, 100, 0); +static const DECLARE_TLV_DB_SCALE(alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(alc_gain_tlv, 0, 600, 0); + +/* ADC and DAC voice mode (8kHz) high pass cutoff value */ +static const char * const da7213_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7213_dac_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, + DA7213_VOICE_HPF_CORNER_MAX, + da7213_voice_hpf_corner_txt); + +static const struct soc_enum da7213_adc_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, + DA7213_VOICE_HPF_CORNER_MAX, + da7213_voice_hpf_corner_txt); + +/* ADC and DAC high pass filter cutoff value */ +static const char * const da7213_audio_hpf_corner_txt[] = { + "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" +}; + +static const struct soc_enum da7213_dac_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, + DA7213_AUDIO_HPF_CORNER_MAX, + da7213_audio_hpf_corner_txt); + +static const struct soc_enum da7213_adc_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, + DA7213_AUDIO_HPF_CORNER_MAX, + da7213_audio_hpf_corner_txt); + +/* Gain ramping rate value */ +static const char * const da7213_gain_ramp_rate_txt[] = { + "nominal rate * 8", "nominal rate * 16", "nominal rate / 16", + "nominal rate / 32" +}; + +static const struct soc_enum da7213_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7213_GAIN_RAMP_CTRL, DA7213_GAIN_RAMP_RATE_SHIFT, + DA7213_GAIN_RAMP_RATE_MAX, da7213_gain_ramp_rate_txt); + +/* DAC noise gate setup time value */ +static const char * const da7213_dac_ng_setup_time_txt[] = { + "256 samples", "512 samples", "1024 samples", "2048 samples" +}; + +static const struct soc_enum da7213_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_SETUP_TIME_SHIFT, + DA7213_DAC_NG_SETUP_TIME_MAX, + da7213_dac_ng_setup_time_txt); + +/* DAC noise gate rampup rate value */ +static const char * const da7213_dac_ng_rampup_txt[] = { + "0.02 ms/dB", "0.16 ms/dB" +}; + +static const struct soc_enum da7213_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPUP_RATE_SHIFT, + DA7213_DAC_NG_RAMP_RATE_MAX, + da7213_dac_ng_rampup_txt); + +/* DAC noise gate rampdown rate value */ +static const char * const da7213_dac_ng_rampdown_txt[] = { + "0.64 ms/dB", "20.48 ms/dB" +}; + +static const struct soc_enum da7213_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPDN_RATE_SHIFT, + DA7213_DAC_NG_RAMP_RATE_MAX, + da7213_dac_ng_rampdown_txt); + +/* DAC soft mute rate value */ +static const char * const da7213_dac_soft_mute_rate_txt[] = { + "1", "2", "4", "8", "16", "32", "64" +}; + +static const struct soc_enum da7213_dac_soft_mute_rate = + SOC_ENUM_SINGLE(DA7213_DAC_FILTERS5, DA7213_DAC_SOFTMUTE_RATE_SHIFT, + DA7213_DAC_SOFTMUTE_RATE_MAX, + da7213_dac_soft_mute_rate_txt); + +/* ALC Attack Rate select */ +static const char * const da7213_alc_attack_rate_txt[] = { + "44/fs", "88/fs", "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", + "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da7213_alc_attack_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_ATTACK_SHIFT, + DA7213_ALC_ATTACK_MAX, da7213_alc_attack_rate_txt); + +/* ALC Release Rate select */ +static const char * const da7213_alc_release_rate_txt[] = { + "176/fs", "352/fs", "704/fs", "1408/fs", "2816/fs", "5632/fs", + "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" +}; + +static const struct soc_enum da7213_alc_release_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_RELEASE_SHIFT, + DA7213_ALC_RELEASE_MAX, da7213_alc_release_rate_txt); + +/* ALC Hold Time select */ +static const char * const da7213_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7213_alc_hold_time = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_HOLD_SHIFT, + DA7213_ALC_HOLD_MAX, da7213_alc_hold_time_txt); + +/* ALC Input Signal Tracking rate select */ +static const char * const da7213_alc_integ_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7213_alc_integ_attack_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_ATTACK_SHIFT, + DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); + +static const struct soc_enum da7213_alc_integ_release_rate = + SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_RELEASE_SHIFT, + DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); + + +/* + * Control Functions + */ + +static int da7213_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) +{ + int mid_data, top_data; + int sum = 0; + u8 iteration; + + for (iteration = 0; iteration < DA7213_ALC_AVG_ITERATIONS; + iteration++) { + /* Select the left or right channel and capture data */ + snd_soc_write(codec, DA7213_ALC_CIC_OP_LVL_CTRL, reg_val); + + /* Select middle 8 bits for read back from data register */ + snd_soc_write(codec, DA7213_ALC_CIC_OP_LVL_CTRL, + reg_val | DA7213_ALC_DATA_MIDDLE); + mid_data = snd_soc_read(codec, DA7213_ALC_CIC_OP_LVL_DATA); + + /* Select top 8 bits for read back from data register */ + snd_soc_write(codec, DA7213_ALC_CIC_OP_LVL_CTRL, + reg_val | DA7213_ALC_DATA_TOP); + top_data = snd_soc_read(codec, DA7213_ALC_CIC_OP_LVL_DATA); + + sum += ((mid_data << 8) | (top_data << 16)); + } + + return sum / DA7213_ALC_AVG_ITERATIONS; +} + +static void da7213_alc_calib_man(struct snd_soc_codec *codec) +{ + u8 reg_val; + int avg_left_data, avg_right_data, offset_l, offset_r; + + /* Calculate average for Left and Right data */ + /* Left Data */ + avg_left_data = da7213_get_alc_data(codec, + DA7213_ALC_CIC_OP_CHANNEL_LEFT); + /* Right Data */ + avg_right_data = da7213_get_alc_data(codec, + DA7213_ALC_CIC_OP_CHANNEL_RIGHT); + + /* Calculate DC offset */ + offset_l = -avg_left_data; + offset_r = -avg_right_data; + + reg_val = (offset_l & DA7213_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_M_L, reg_val); + reg_val = (offset_l & DA7213_ALC_OFFSET_19_16) >> 16; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_U_L, reg_val); + + reg_val = (offset_r & DA7213_ALC_OFFSET_15_8) >> 8; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_M_R, reg_val); + reg_val = (offset_r & DA7213_ALC_OFFSET_19_16) >> 16; + snd_soc_write(codec, DA7213_ALC_OFFSET_MAN_U_R, reg_val); + + /* Enable analog/digital gain mode & offset cancellation */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE); +} + +static void da7213_alc_calib_auto(struct snd_soc_codec *codec) +{ + u8 alc_ctrl1; + + /* Begin auto calibration and wait for completion */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, DA7213_ALC_AUTO_CALIB_EN, + DA7213_ALC_AUTO_CALIB_EN); + do { + alc_ctrl1 = snd_soc_read(codec, DA7213_ALC_CTRL1); + } while (alc_ctrl1 & DA7213_ALC_AUTO_CALIB_EN); + + /* If auto calibration fails, fall back to digital gain only mode */ + if (alc_ctrl1 & DA7213_ALC_CALIB_OVERFLOW) { + dev_warn(codec->dev, + "ALC auto calibration failed with overflow\n"); + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE, + 0); + } else { + /* Enable analog/digital gain mode & offset cancellation */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE, + DA7213_ALC_OFFSET_EN | DA7213_ALC_SYNC_MODE); + } + +} + +static void da7213_alc_calib(struct snd_soc_codec *codec) +{ + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 adc_l_ctrl, adc_r_ctrl; + u8 mixin_l_sel, mixin_r_sel; + u8 mic_1_ctrl, mic_2_ctrl; + + /* Save current values from ADC control registers */ + adc_l_ctrl = snd_soc_read(codec, DA7213_ADC_L_CTRL); + adc_r_ctrl = snd_soc_read(codec, DA7213_ADC_R_CTRL); + + /* Save current values from MIXIN_L/R_SELECT registers */ + mixin_l_sel = snd_soc_read(codec, DA7213_MIXIN_L_SELECT); + mixin_r_sel = snd_soc_read(codec, DA7213_MIXIN_R_SELECT); + + /* Save current values from MIC control registers */ + mic_1_ctrl = snd_soc_read(codec, DA7213_MIC_1_CTRL); + mic_2_ctrl = snd_soc_read(codec, DA7213_MIC_2_CTRL); + + /* Enable ADC Left and Right */ + snd_soc_update_bits(codec, DA7213_ADC_L_CTRL, DA7213_ADC_EN, + DA7213_ADC_EN); + snd_soc_update_bits(codec, DA7213_ADC_R_CTRL, DA7213_ADC_EN, + DA7213_ADC_EN); + + /* Enable MIC paths */ + snd_soc_update_bits(codec, DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIC_1 | + DA7213_MIXIN_L_MIX_SELECT_MIC_2, + DA7213_MIXIN_L_MIX_SELECT_MIC_1 | + DA7213_MIXIN_L_MIX_SELECT_MIC_2); + snd_soc_update_bits(codec, DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIC_2 | + DA7213_MIXIN_R_MIX_SELECT_MIC_1, + DA7213_MIXIN_R_MIX_SELECT_MIC_2 | + DA7213_MIXIN_R_MIX_SELECT_MIC_1); + + /* Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7213_MIC_1_CTRL, DA7213_MUTE_EN, + DA7213_MUTE_EN); + snd_soc_update_bits(codec, DA7213_MIC_2_CTRL, DA7213_MUTE_EN, + DA7213_MUTE_EN); + + /* Perform calibration */ + if (da7213->alc_calib_auto) + da7213_alc_calib_auto(codec); + else + da7213_alc_calib_man(codec); + + /* Restore MIXIN_L/R_SELECT registers to their original states */ + snd_soc_write(codec, DA7213_MIXIN_L_SELECT, mixin_l_sel); + snd_soc_write(codec, DA7213_MIXIN_R_SELECT, mixin_r_sel); + + /* Restore ADC control registers to their original states */ + snd_soc_write(codec, DA7213_ADC_L_CTRL, adc_l_ctrl); + snd_soc_write(codec, DA7213_ADC_R_CTRL, adc_r_ctrl); + + /* Restore original values of MIC control registers */ + snd_soc_write(codec, DA7213_MIC_1_CTRL, mic_1_ctrl); + snd_soc_write(codec, DA7213_MIC_2_CTRL, mic_2_ctrl); +} + +static int da7213_put_mixin_gain(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + + /* If ALC in operation, make sure calibrated offsets are updated */ + if ((!ret) && (da7213->alc_en)) + da7213_alc_calib(codec); + + return ret; +} + +static int da7213_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + /* Force ALC offset calibration if enabling ALC */ + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) { + if (!da7213->alc_en) { + da7213_alc_calib(codec); + da7213->alc_en = true; + } + } else { + da7213->alc_en = false; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7213_snd_controls[] = { + + /* Volume controls */ + SOC_SINGLE_TLV("Mic 1 Volume", DA7213_MIC_1_GAIN, + DA7213_MIC_AMP_GAIN_SHIFT, DA7213_MIC_AMP_GAIN_MAX, + DA7213_NO_INVERT, mic_vol_tlv), + SOC_SINGLE_TLV("Mic 2 Volume", DA7213_MIC_2_GAIN, + DA7213_MIC_AMP_GAIN_SHIFT, DA7213_MIC_AMP_GAIN_MAX, + DA7213_NO_INVERT, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux Volume", DA7213_AUX_L_GAIN, DA7213_AUX_R_GAIN, + DA7213_AUX_AMP_GAIN_SHIFT, DA7213_AUX_AMP_GAIN_MAX, + DA7213_NO_INVERT, aux_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("Mixin PGA Volume", DA7213_MIXIN_L_GAIN, + DA7213_MIXIN_R_GAIN, DA7213_MIXIN_AMP_GAIN_SHIFT, + DA7213_MIXIN_AMP_GAIN_MAX, DA7213_NO_INVERT, + snd_soc_get_volsw_2r, da7213_put_mixin_gain, + mixin_gain_tlv), + SOC_DOUBLE_R_TLV("ADC Volume", DA7213_ADC_L_GAIN, DA7213_ADC_R_GAIN, + DA7213_ADC_AMP_GAIN_SHIFT, DA7213_ADC_AMP_GAIN_MAX, + DA7213_NO_INVERT, digital_gain_tlv), + SOC_DOUBLE_R_TLV("DAC Volume", DA7213_DAC_L_GAIN, DA7213_DAC_R_GAIN, + DA7213_DAC_AMP_GAIN_SHIFT, DA7213_DAC_AMP_GAIN_MAX, + DA7213_NO_INVERT, digital_gain_tlv), + SOC_DOUBLE_R_TLV("Headphone Volume", DA7213_HP_L_GAIN, DA7213_HP_R_GAIN, + DA7213_HP_AMP_GAIN_SHIFT, DA7213_HP_AMP_GAIN_MAX, + DA7213_NO_INVERT, hp_vol_tlv), + SOC_SINGLE_TLV("Lineout Volume", DA7213_LINE_GAIN, + DA7213_LINE_AMP_GAIN_SHIFT, DA7213_LINE_AMP_GAIN_MAX, + DA7213_NO_INVERT, lineout_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7213_DAC_FILTERS4, DA7213_DAC_EQ_EN_SHIFT, + DA7213_DAC_EQ_EN_MAX, DA7213_NO_INVERT), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7213_DAC_FILTERS2, + DA7213_DAC_EQ_BAND1_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7213_DAC_FILTERS2, + DA7213_DAC_EQ_BAND2_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7213_DAC_FILTERS3, + DA7213_DAC_EQ_BAND3_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7213_DAC_FILTERS3, + DA7213_DAC_EQ_BAND4_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7213_DAC_FILTERS4, + DA7213_DAC_EQ_BAND5_SHIFT, DA7213_DAC_EQ_BAND_MAX, + DA7213_NO_INVERT, eq_gain_tlv), + + /* High Pass Filter and Voice Mode controls */ + SOC_SINGLE("ADC HPF Switch", DA7213_ADC_FILTERS1, DA7213_HPF_EN_SHIFT, + DA7213_HPF_EN_MAX, DA7213_NO_INVERT), + SOC_ENUM("ADC HPF Cutoff", da7213_adc_audio_hpf_corner), + SOC_SINGLE("ADC Voice Mode Switch", DA7213_ADC_FILTERS1, + DA7213_VOICE_EN_SHIFT, DA7213_VOICE_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("ADC Voice Cutoff", da7213_adc_voice_hpf_corner), + + SOC_SINGLE("DAC HPF Switch", DA7213_DAC_FILTERS1, DA7213_HPF_EN_SHIFT, + DA7213_HPF_EN_MAX, DA7213_NO_INVERT), + SOC_ENUM("DAC HPF Cutoff", da7213_dac_audio_hpf_corner), + SOC_SINGLE("DAC Voice Mode Switch", DA7213_DAC_FILTERS1, + DA7213_VOICE_EN_SHIFT, DA7213_VOICE_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("DAC Voice Cutoff", da7213_dac_voice_hpf_corner), + + /* Mute controls */ + SOC_SINGLE("Mic 1 Switch", DA7213_MIC_1_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_SINGLE("Mic 2 Switch", DA7213_MIC_2_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("Aux Switch", DA7213_AUX_L_CTRL, DA7213_AUX_R_CTRL, + DA7213_MUTE_EN_SHIFT, DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("Mixin PGA Switch", DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_R_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("ADC Switch", DA7213_ADC_L_CTRL, DA7213_ADC_R_CTRL, + DA7213_MUTE_EN_SHIFT, DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_DOUBLE_R("Headphone Switch", DA7213_HP_L_CTRL, DA7213_HP_R_CTRL, + DA7213_MUTE_EN_SHIFT, DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_SINGLE("Lineout Switch", DA7213_LINE_CTRL, DA7213_MUTE_EN_SHIFT, + DA7213_MUTE_EN_MAX, DA7213_INVERT), + SOC_SINGLE("DAC Soft Mute Switch", DA7213_DAC_FILTERS5, + DA7213_DAC_SOFTMUTE_EN_SHIFT, DA7213_DAC_SOFTMUTE_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("DAC Soft Mute Rate", da7213_dac_soft_mute_rate), + + /* Zero Cross controls */ + SOC_DOUBLE_R("Aux ZC Switch", DA7213_AUX_L_CTRL, DA7213_AUX_R_CTRL, + DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("Mixin PGA ZC Switch", DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_R_CTRL, DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, + DA7213_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Switch", DA7213_HP_L_CTRL, DA7213_HP_R_CTRL, + DA7213_ZC_EN_SHIFT, DA7213_ZC_EN_MAX, DA7213_NO_INVERT), + + /* Gain Ramping controls */ + SOC_DOUBLE_R("Aux Gain Ramping Switch", DA7213_AUX_L_CTRL, + DA7213_AUX_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("Mixin Gain Ramping Switch", DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("ADC Gain Ramping Switch", DA7213_ADC_L_CTRL, + DA7213_ADC_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("DAC Gain Ramping Switch", DA7213_DAC_L_CTRL, + DA7213_DAC_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_DOUBLE_R("Headphone Gain Ramping Switch", DA7213_HP_L_CTRL, + DA7213_HP_R_CTRL, DA7213_GAIN_RAMP_EN_SHIFT, + DA7213_GAIN_RAMP_EN_MAX, DA7213_NO_INVERT), + SOC_SINGLE("Lineout Gain Ramping Switch", DA7213_LINE_CTRL, + DA7213_GAIN_RAMP_EN_SHIFT, DA7213_GAIN_RAMP_EN_MAX, + DA7213_NO_INVERT), + SOC_ENUM("Gain Ramping Rate", da7213_gain_ramp_rate), + + /* DAC Noise Gate controls */ + SOC_SINGLE("DAC NG Switch", DA7213_DAC_NG_CTRL, DA7213_DAC_NG_EN_SHIFT, + DA7213_DAC_NG_EN_MAX, DA7213_NO_INVERT), + SOC_ENUM("DAC NG Setup Time", da7213_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7213_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7213_dac_ng_rampdown_rate), + SOC_SINGLE("DAC NG OFF Threshold", DA7213_DAC_NG_OFF_THRESHOLD, + DA7213_DAC_NG_THRESHOLD_SHIFT, DA7213_DAC_NG_THRESHOLD_MAX, + DA7213_NO_INVERT), + SOC_SINGLE("DAC NG ON Threshold", DA7213_DAC_NG_ON_THRESHOLD, + DA7213_DAC_NG_THRESHOLD_SHIFT, DA7213_DAC_NG_THRESHOLD_MAX, + DA7213_NO_INVERT), + + /* DAC Routing & Inversion */ + SOC_DOUBLE("DAC Mono Switch", DA7213_DIG_ROUTING_DAC, + DA7213_DAC_L_MONO_SHIFT, DA7213_DAC_R_MONO_SHIFT, + DA7213_DAC_MONO_MAX, DA7213_NO_INVERT), + SOC_DOUBLE("DAC Invert Switch", DA7213_DIG_CTRL, DA7213_DAC_L_INV_SHIFT, + DA7213_DAC_R_INV_SHIFT, DA7213_DAC_INV_MAX, + DA7213_NO_INVERT), + + /* DMIC controls */ + SOC_DOUBLE_R("DMIC Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_R_SELECT, DA7213_DMIC_EN_SHIFT, + DA7213_DMIC_EN_MAX, DA7213_NO_INVERT), + + /* ALC Controls */ + SOC_DOUBLE_EXT("ALC Switch", DA7213_ALC_CTRL1, DA7213_ALC_L_EN_SHIFT, + DA7213_ALC_R_EN_SHIFT, DA7213_ALC_EN_MAX, + DA7213_NO_INVERT, snd_soc_get_volsw, da7213_put_alc_sw), + SOC_ENUM("ALC Attack Rate", da7213_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7213_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7213_alc_hold_time), + /* + * Rate at which input signal envelope is tracked as the signal gets + * larger + */ + SOC_ENUM("ALC Integ Attack Rate", da7213_alc_integ_attack_rate), + /* + * Rate at which input signal envelope is tracked as the signal gets + * smaller + */ + SOC_ENUM("ALC Integ Release Rate", da7213_alc_integ_release_rate), + SOC_SINGLE_TLV("ALC Noise Threshold Volume", DA7213_ALC_NOISE, + DA7213_ALC_THRESHOLD_SHIFT, DA7213_ALC_THRESHOLD_MAX, + DA7213_INVERT, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", DA7213_ALC_TARGET_MIN, + DA7213_ALC_THRESHOLD_SHIFT, DA7213_ALC_THRESHOLD_MAX, + DA7213_INVERT, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold Volume", DA7213_ALC_TARGET_MAX, + DA7213_ALC_THRESHOLD_SHIFT, DA7213_ALC_THRESHOLD_MAX, + DA7213_INVERT, alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation Volume", DA7213_ALC_GAIN_LIMITS, + DA7213_ALC_ATTEN_MAX_SHIFT, + DA7213_ALC_ATTEN_GAIN_MAX_MAX, DA7213_NO_INVERT, + alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Gain Volume", DA7213_ALC_GAIN_LIMITS, + DA7213_ALC_GAIN_MAX_SHIFT, DA7213_ALC_ATTEN_GAIN_MAX_MAX, + DA7213_NO_INVERT, alc_gain_tlv), + SOC_SINGLE_TLV("ALC Min Analog Gain Volume", DA7213_ALC_ANA_GAIN_LIMITS, + DA7213_ALC_ANA_GAIN_MIN_SHIFT, DA7213_ALC_ANA_GAIN_MAX, + DA7213_NO_INVERT, alc_analog_gain_tlv), + SOC_SINGLE_TLV("ALC Max Analog Gain Volume", DA7213_ALC_ANA_GAIN_LIMITS, + DA7213_ALC_ANA_GAIN_MAX_SHIFT, DA7213_ALC_ANA_GAIN_MAX, + DA7213_NO_INVERT, alc_analog_gain_tlv), + SOC_SINGLE("ALC Anticlip Mode Switch", DA7213_ALC_ANTICLIP_CTRL, + DA7213_ALC_ANTICLIP_EN_SHIFT, DA7213_ALC_ANTICLIP_EN_MAX, + DA7213_NO_INVERT), + SOC_SINGLE("ALC Anticlip Level", DA7213_ALC_ANTICLIP_LEVEL, + DA7213_ALC_ANTICLIP_LEVEL_SHIFT, + DA7213_ALC_ANTICLIP_LEVEL_MAX, DA7213_NO_INVERT), +}; + + +/* + * DAPM + */ + +/* + * Enums + */ + +/* MIC PGA source select */ +static const char * const da7213_mic_amp_in_sel_txt[] = { + "Differential", "MIC_P", "MIC_N" +}; + +static const struct soc_enum da7213_mic_1_amp_in_sel = + SOC_ENUM_SINGLE(DA7213_MIC_1_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, + DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static const struct snd_kcontrol_new da7213_mic_1_amp_in_sel_mux = + SOC_DAPM_ENUM("Mic 1 Amp Source MUX", da7213_mic_1_amp_in_sel); + +static const struct soc_enum da7213_mic_2_amp_in_sel = + SOC_ENUM_SINGLE(DA7213_MIC_2_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, + DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static const struct snd_kcontrol_new da7213_mic_2_amp_in_sel_mux = + SOC_DAPM_ENUM("Mic 2 Amp Source MUX", da7213_mic_2_amp_in_sel); + +/* DAI routing select */ +static const char * const da7213_dai_src_txt[] = { + "ADC Left", "ADC Right", "DAI Input Left", "DAI Input Right" +}; + +static const struct soc_enum da7213_dai_l_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_L_SRC_SHIFT, + DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static const struct snd_kcontrol_new da7213_dai_l_src_mux = + SOC_DAPM_ENUM("DAI Left Source MUX", da7213_dai_l_src); + +static const struct soc_enum da7213_dai_r_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_R_SRC_SHIFT, + DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static const struct snd_kcontrol_new da7213_dai_r_src_mux = + SOC_DAPM_ENUM("DAI Right Source MUX", da7213_dai_r_src); + +/* DAC routing select */ +static const char * const da7213_dac_src_txt[] = { + "ADC Output Left", "ADC Output Right", "DAI Input Left", + "DAI Input Right" +}; + +static const struct soc_enum da7213_dac_l_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_L_SRC_SHIFT, + DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static const struct snd_kcontrol_new da7213_dac_l_src_mux = + SOC_DAPM_ENUM("DAC Left Source MUX", da7213_dac_l_src); + +static const struct soc_enum da7213_dac_r_src = + SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_R_SRC_SHIFT, + DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static const struct snd_kcontrol_new da7213_dac_r_src_mux = + SOC_DAPM_ENUM("DAC Right Source MUX", da7213_dac_r_src); + +/* + * Mixer Controls + */ + +/* Mixin Left */ +static const struct snd_kcontrol_new da7213_dapm_mixinl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_AUX_L_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 1 Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIC_1_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 2 Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIC_2_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Switch", DA7213_MIXIN_L_SELECT, + DA7213_MIXIN_L_MIX_SELECT_MIXIN_R_SHIFT, + DA7213_MIXIN_L_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + +/* Mixin Right */ +static const struct snd_kcontrol_new da7213_dapm_mixinr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_AUX_R_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 2 Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIC_2_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mic 1 Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIC_1_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Switch", DA7213_MIXIN_R_SELECT, + DA7213_MIXIN_R_MIX_SELECT_MIXIN_L_SHIFT, + DA7213_MIXIN_R_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + +/* Mixout Left */ +static const struct snd_kcontrol_new da7213_dapm_mixoutl_controls[] = { + SOC_DAPM_SINGLE("Aux Left Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_AUX_L_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("DAC Left Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_DAC_L_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Aux Left Invert Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_AUX_L_INVERTED_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_INVERTED_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA7213_MIXOUT_L_SELECT, + DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_INVERTED_SHIFT, + DA7213_MIXOUT_L_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + +/* Mixout Right */ +static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = { + SOC_DAPM_SINGLE("Aux Right Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_AUX_R_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("DAC Right Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_DAC_R_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Aux Right Invert Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_AUX_R_INVERTED_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Right Invert Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_INVERTED_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), + SOC_DAPM_SINGLE("Mixin Left Invert Switch", DA7213_MIXOUT_R_SELECT, + DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_INVERTED_SHIFT, + DA7213_MIXOUT_R_MIX_SELECT_MAX, DA7213_NO_INVERT), +}; + + +/* + * DAPM widgets + */ + +static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = { + /* + * Input & Output + */ + + /* Use a supply here as this controls both input & output DAIs */ + SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + + /* + * Input + */ + + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + /* MUXs for Mic PGA source selection */ + SND_SOC_DAPM_MUX("Mic 1 Amp Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_mic_1_amp_in_sel_mux), + SND_SOC_DAPM_MUX("Mic 2 Amp Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_mic_2_amp_in_sel_mux), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic 1 PGA", DA7213_MIC_1_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mic 2 PGA", DA7213_MIC_2_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Aux Left PGA", DA7213_AUX_L_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Aux Right PGA", DA7213_AUX_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mixin Left PGA", DA7213_MIXIN_L_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mixin Right PGA", DA7213_MIXIN_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + + /* Mic Biases */ + SND_SOC_DAPM_SUPPLY("Mic Bias 1", DA7213_MICBIAS_CTRL, + DA7213_MICBIAS1_EN_SHIFT, DA7213_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias 2", DA7213_MICBIAS_CTRL, + DA7213_MICBIAS2_EN_SHIFT, DA7213_NO_INVERT, + NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("Mixin Left", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixinl_controls[0], + ARRAY_SIZE(da7213_dapm_mixinl_controls)), + SND_SOC_DAPM_MIXER("Mixin Right", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixinr_controls[0], + ARRAY_SIZE(da7213_dapm_mixinr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", NULL, DA7213_ADC_L_CTRL, + DA7213_ADC_EN_SHIFT, DA7213_NO_INVERT), + SND_SOC_DAPM_ADC("ADC Right", NULL, DA7213_ADC_R_CTRL, + DA7213_ADC_EN_SHIFT, DA7213_NO_INVERT), + + /* DAI */ + SND_SOC_DAPM_MUX("DAI Left Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dai_l_src_mux), + SND_SOC_DAPM_MUX("DAI Right Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dai_r_src_mux), + SND_SOC_DAPM_AIF_OUT("DAIOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DAIOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + + /* + * Output + */ + + /* DAI */ + SND_SOC_DAPM_AIF_IN("DAIINL", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("DAC Left Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dac_l_src_mux), + SND_SOC_DAPM_MUX("DAC Right Source MUX", SND_SOC_NOPM, 0, 0, + &da7213_dac_r_src_mux), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", NULL, DA7213_DAC_L_CTRL, + DA7213_DAC_EN_SHIFT, DA7213_NO_INVERT), + SND_SOC_DAPM_DAC("DAC Right", NULL, DA7213_DAC_R_CTRL, + DA7213_DAC_EN_SHIFT, DA7213_NO_INVERT), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixout Left", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixoutl_controls[0], + ARRAY_SIZE(da7213_dapm_mixoutl_controls)), + SND_SOC_DAPM_MIXER("Mixout Right", SND_SOC_NOPM, 0, 0, + &da7213_dapm_mixoutr_controls[0], + ARRAY_SIZE(da7213_dapm_mixoutr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7213_MIXOUT_L_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7213_MIXOUT_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Lineout PGA", DA7213_LINE_CTRL, DA7213_AMP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left PGA", DA7213_HP_L_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right PGA", DA7213_HP_R_CTRL, + DA7213_AMP_EN_SHIFT, DA7213_NO_INVERT, NULL, 0), + + /* Charge Pump */ + SND_SOC_DAPM_SUPPLY("Charge Pump", DA7213_CP_CTRL, DA7213_CP_EN_SHIFT, + DA7213_NO_INVERT, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LINE"), +}; + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7213_audio_map[] = { + /* Dest Connecting Widget source */ + + /* Input path */ + {"MIC1", NULL, "Mic Bias 1"}, + {"MIC2", NULL, "Mic Bias 2"}, + + {"Mic 1 Amp Source MUX", "Differential", "MIC1"}, + {"Mic 1 Amp Source MUX", "MIC_P", "MIC1"}, + {"Mic 1 Amp Source MUX", "MIC_N", "MIC1"}, + + {"Mic 2 Amp Source MUX", "Differential", "MIC2"}, + {"Mic 2 Amp Source MUX", "MIC_P", "MIC2"}, + {"Mic 2 Amp Source MUX", "MIC_N", "MIC2"}, + + {"Mic 1 PGA", NULL, "Mic 1 Amp Source MUX"}, + {"Mic 2 PGA", NULL, "Mic 2 Amp Source MUX"}, + + {"Aux Left PGA", NULL, "AUXL"}, + {"Aux Right PGA", NULL, "AUXR"}, + + {"Mixin Left", "Aux Left Switch", "Aux Left PGA"}, + {"Mixin Left", "Mic 1 Switch", "Mic 1 PGA"}, + {"Mixin Left", "Mic 2 Switch", "Mic 2 PGA"}, + {"Mixin Left", "Mixin Right Switch", "Mixin Right PGA"}, + + {"Mixin Right", "Aux Right Switch", "Aux Right PGA"}, + {"Mixin Right", "Mic 2 Switch", "Mic 2 PGA"}, + {"Mixin Right", "Mic 1 Switch", "Mic 1 PGA"}, + {"Mixin Right", "Mixin Left Switch", "Mixin Left PGA"}, + + {"Mixin Left PGA", NULL, "Mixin Left"}, + {"ADC Left", NULL, "Mixin Left PGA"}, + + {"Mixin Right PGA", NULL, "Mixin Right"}, + {"ADC Right", NULL, "Mixin Right PGA"}, + + {"DAI Left Source MUX", "ADC Left", "ADC Left"}, + {"DAI Left Source MUX", "ADC Right", "ADC Right"}, + {"DAI Left Source MUX", "DAI Input Left", "DAIINL"}, + {"DAI Left Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAI Right Source MUX", "ADC Left", "ADC Left"}, + {"DAI Right Source MUX", "ADC Right", "ADC Right"}, + {"DAI Right Source MUX", "DAI Input Left", "DAIINL"}, + {"DAI Right Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAIOUTL", NULL, "DAI Left Source MUX"}, + {"DAIOUTR", NULL, "DAI Right Source MUX"}, + + {"DAIOUTL", NULL, "DAI"}, + {"DAIOUTR", NULL, "DAI"}, + + /* Output path */ + {"DAIINL", NULL, "DAI"}, + {"DAIINR", NULL, "DAI"}, + + {"DAC Left Source MUX", "ADC Output Left", "ADC Left"}, + {"DAC Left Source MUX", "ADC Output Right", "ADC Right"}, + {"DAC Left Source MUX", "DAI Input Left", "DAIINL"}, + {"DAC Left Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAC Right Source MUX", "ADC Output Left", "ADC Left"}, + {"DAC Right Source MUX", "ADC Output Right", "ADC Right"}, + {"DAC Right Source MUX", "DAI Input Left", "DAIINL"}, + {"DAC Right Source MUX", "DAI Input Right", "DAIINR"}, + + {"DAC Left", NULL, "DAC Left Source MUX"}, + {"DAC Right", NULL, "DAC Right Source MUX"}, + + {"Mixout Left", "Aux Left Switch", "Aux Left PGA"}, + {"Mixout Left", "Mixin Left Switch", "Mixin Left PGA"}, + {"Mixout Left", "Mixin Right Switch", "Mixin Right PGA"}, + {"Mixout Left", "DAC Left Switch", "DAC Left"}, + {"Mixout Left", "Aux Left Invert Switch", "Aux Left PGA"}, + {"Mixout Left", "Mixin Left Invert Switch", "Mixin Left PGA"}, + {"Mixout Left", "Mixin Right Invert Switch", "Mixin Right PGA"}, + + {"Mixout Right", "Aux Right Switch", "Aux Right PGA"}, + {"Mixout Right", "Mixin Right Switch", "Mixin Right PGA"}, + {"Mixout Right", "Mixin Left Switch", "Mixin Left PGA"}, + {"Mixout Right", "DAC Right Switch", "DAC Right"}, + {"Mixout Right", "Aux Right Invert Switch", "Aux Right PGA"}, + {"Mixout Right", "Mixin Right Invert Switch", "Mixin Right PGA"}, + {"Mixout Right", "Mixin Left Invert Switch", "Mixin Left PGA"}, + + {"Mixout Left PGA", NULL, "Mixout Left"}, + {"Mixout Right PGA", NULL, "Mixout Right"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Left PGA", NULL, "Charge Pump"}, + {"HPL", NULL, "Headphone Left PGA"}, + + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + {"Headphone Right PGA", NULL, "Charge Pump"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"Lineout PGA", NULL, "Mixout Right PGA"}, + {"LINE", NULL, "Lineout PGA"}, +}; + +static struct reg_default da7213_reg_defaults[] = { + { DA7213_DIG_ROUTING_DAI, 0x10 }, + { DA7213_SR, 0x0A }, + { DA7213_REFERENCES, 0x80 }, + { DA7213_PLL_FRAC_TOP, 0x00 }, + { DA7213_PLL_FRAC_BOT, 0x00 }, + { DA7213_PLL_INTEGER, 0x20 }, + { DA7213_PLL_CTRL, 0x0C }, + { DA7213_DAI_CLK_MODE, 0x01 }, + { DA7213_DAI_CTRL, 0x08 }, + { DA7213_DIG_ROUTING_DAC, 0x32 }, + { DA7213_AUX_L_GAIN, 0x35 }, + { DA7213_AUX_R_GAIN, 0x35 }, + { DA7213_MIXIN_L_SELECT, 0x00 }, + { DA7213_MIXIN_R_SELECT, 0x00 }, + { DA7213_MIXIN_L_GAIN, 0x03 }, + { DA7213_MIXIN_R_GAIN, 0x03 }, + { DA7213_ADC_L_GAIN, 0x6F }, + { DA7213_ADC_R_GAIN, 0x6F }, + { DA7213_ADC_FILTERS1, 0x80 }, + { DA7213_MIC_1_GAIN, 0x01 }, + { DA7213_MIC_2_GAIN, 0x01 }, + { DA7213_DAC_FILTERS5, 0x00 }, + { DA7213_DAC_FILTERS2, 0x88 }, + { DA7213_DAC_FILTERS3, 0x88 }, + { DA7213_DAC_FILTERS4, 0x08 }, + { DA7213_DAC_FILTERS1, 0x80 }, + { DA7213_DAC_L_GAIN, 0x6F }, + { DA7213_DAC_R_GAIN, 0x6F }, + { DA7213_CP_CTRL, 0x61 }, + { DA7213_HP_L_GAIN, 0x39 }, + { DA7213_HP_R_GAIN, 0x39 }, + { DA7213_LINE_GAIN, 0x30 }, + { DA7213_MIXOUT_L_SELECT, 0x00 }, + { DA7213_MIXOUT_R_SELECT, 0x00 }, + { DA7213_SYSTEM_MODES_INPUT, 0x00 }, + { DA7213_SYSTEM_MODES_OUTPUT, 0x00 }, + { DA7213_AUX_L_CTRL, 0x44 }, + { DA7213_AUX_R_CTRL, 0x44 }, + { DA7213_MICBIAS_CTRL, 0x11 }, + { DA7213_MIC_1_CTRL, 0x40 }, + { DA7213_MIC_2_CTRL, 0x40 }, + { DA7213_MIXIN_L_CTRL, 0x40 }, + { DA7213_MIXIN_R_CTRL, 0x40 }, + { DA7213_ADC_L_CTRL, 0x40 }, + { DA7213_ADC_R_CTRL, 0x40 }, + { DA7213_DAC_L_CTRL, 0x48 }, + { DA7213_DAC_R_CTRL, 0x40 }, + { DA7213_HP_L_CTRL, 0x41 }, + { DA7213_HP_R_CTRL, 0x40 }, + { DA7213_LINE_CTRL, 0x40 }, + { DA7213_MIXOUT_L_CTRL, 0x10 }, + { DA7213_MIXOUT_R_CTRL, 0x10 }, + { DA7213_LDO_CTRL, 0x00 }, + { DA7213_IO_CTRL, 0x00 }, + { DA7213_GAIN_RAMP_CTRL, 0x00}, + { DA7213_MIC_CONFIG, 0x00 }, + { DA7213_PC_COUNT, 0x00 }, + { DA7213_CP_VOL_THRESHOLD1, 0x32 }, + { DA7213_CP_DELAY, 0x95 }, + { DA7213_CP_DETECTOR, 0x00 }, + { DA7213_DAI_OFFSET, 0x00 }, + { DA7213_DIG_CTRL, 0x00 }, + { DA7213_ALC_CTRL2, 0x00 }, + { DA7213_ALC_CTRL3, 0x00 }, + { DA7213_ALC_NOISE, 0x3F }, + { DA7213_ALC_TARGET_MIN, 0x3F }, + { DA7213_ALC_TARGET_MAX, 0x00 }, + { DA7213_ALC_GAIN_LIMITS, 0xFF }, + { DA7213_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7213_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7213_ALC_ANTICLIP_LEVEL, 0x00 }, + { DA7213_ALC_OFFSET_MAN_M_L, 0x00 }, + { DA7213_ALC_OFFSET_MAN_U_L, 0x00 }, + { DA7213_ALC_OFFSET_MAN_M_R, 0x00 }, + { DA7213_ALC_OFFSET_MAN_U_R, 0x00 }, + { DA7213_ALC_CIC_OP_LVL_CTRL, 0x00 }, + { DA7213_DAC_NG_SETUP_TIME, 0x00 }, + { DA7213_DAC_NG_OFF_THRESHOLD, 0x00 }, + { DA7213_DAC_NG_ON_THRESHOLD, 0x00 }, + { DA7213_DAC_NG_CTRL, 0x00 }, +}; + +static bool da7213_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7213_STATUS1: + case DA7213_PLL_STATUS: + case DA7213_AUX_L_GAIN_STATUS: + case DA7213_AUX_R_GAIN_STATUS: + case DA7213_MIC_1_GAIN_STATUS: + case DA7213_MIC_2_GAIN_STATUS: + case DA7213_MIXIN_L_GAIN_STATUS: + case DA7213_MIXIN_R_GAIN_STATUS: + case DA7213_ADC_L_GAIN_STATUS: + case DA7213_ADC_R_GAIN_STATUS: + case DA7213_DAC_L_GAIN_STATUS: + case DA7213_DAC_R_GAIN_STATUS: + case DA7213_HP_L_GAIN_STATUS: + case DA7213_HP_R_GAIN_STATUS: + case DA7213_LINE_GAIN_STATUS: + case DA7213_ALC_CTRL1: + case DA7213_ALC_OFFSET_AUTO_M_L: + case DA7213_ALC_OFFSET_AUTO_U_L: + case DA7213_ALC_OFFSET_AUTO_M_R: + case DA7213_ALC_OFFSET_AUTO_U_R: + case DA7213_ALC_CIC_OP_LVL_DATA: + return 1; + default: + return 0; + } +} + +static int da7213_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dai_ctrl = 0; + u8 fs; + + /* Set DAI format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S20_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_ctrl |= DA7213_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + /* Set sampling rate */ + switch (params_rate(params)) { + case 8000: + fs = DA7213_SR_8000; + break; + case 11025: + fs = DA7213_SR_11025; + break; + case 12000: + fs = DA7213_SR_12000; + break; + case 16000: + fs = DA7213_SR_16000; + break; + case 22050: + fs = DA7213_SR_22050; + break; + case 32000: + fs = DA7213_SR_32000; + break; + case 44100: + fs = DA7213_SR_44100; + break; + case 48000: + fs = DA7213_SR_48000; + break; + case 88200: + fs = DA7213_SR_88200; + break; + case 96000: + fs = DA7213_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_WORD_LENGTH_MASK, + dai_ctrl); + snd_soc_write(codec, DA7213_SR, fs); + + return 0; +} + +static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + /* Set master/slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE; + da7213->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE; + da7213->master = false; + break; + default: + return -EINVAL; + } + + /* Set clock normal/inverted */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7213_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7213_DAI_WCLK_POL_INV | DA7213_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + /* Only I2S is supported */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7213_DAI_FORMAT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7213_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7213_DAI_FORMAT_RIGHT_J; + break; + default: + return -EINVAL; + } + + /* By default only 32 BCLK per WCLK is supported */ + dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_32; + + snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode); + snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7213_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) { + snd_soc_update_bits(codec, DA7213_DAC_L_CTRL, + DA7213_MUTE_EN, DA7213_MUTE_EN); + snd_soc_update_bits(codec, DA7213_DAC_R_CTRL, + DA7213_MUTE_EN, DA7213_MUTE_EN); + } else { + snd_soc_update_bits(codec, DA7213_DAC_L_CTRL, + DA7213_MUTE_EN, 0); + snd_soc_update_bits(codec, DA7213_DAC_R_CTRL, + DA7213_MUTE_EN, 0); + } + + return 0; +} + +#define DA7213_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static int da7213_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7213_CLKSRC_MCLK: + if ((freq == 32768) || + ((freq >= 5000000) && (freq <= 54000000))) { + da7213->mclk_rate = freq; + return 0; + } else { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/* Supported PLL input frequencies are 5MHz - 54MHz. */ +static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Reset PLL configuration */ + snd_soc_write(codec, DA7213_PLL_CTRL, 0); + + pll_ctrl = 0; + + /* Workout input divider based on MCLK rate */ + if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) { + /* 32KHz PLL Mode */ + indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; + indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + freq_ref = 3750000; + pll_ctrl |= DA7213_PLL_32K_MODE; + } else { + /* 5 - 54MHz MCLK */ + if (da7213->mclk_rate < 5000000) { + goto pll_err; + } else if (da7213->mclk_rate <= 10000000) { + indiv_bits = DA7213_PLL_INDIV_5_10_MHZ; + indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL; + } else if (da7213->mclk_rate <= 20000000) { + indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; + indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + } else if (da7213->mclk_rate <= 40000000) { + indiv_bits = DA7213_PLL_INDIV_20_40_MHZ; + indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7213->mclk_rate <= 54000000) { + indiv_bits = DA7213_PLL_INDIV_40_54_MHZ; + indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL; + } else { + goto pll_err; + } + freq_ref = (da7213->mclk_rate / indiv); + } + + pll_ctrl |= indiv_bits; + + /* PLL Bypass mode */ + if (source == DA7213_SYSCLK_MCLK) { + snd_soc_write(codec, DA7213_PLL_CTRL, pll_ctrl); + return 0; + } + + /* + * If Codec is slave and SRM enabled, + * freq_out is (98304000 + 90316800)/2 = 94310400 + */ + if (!da7213->master && da7213->srm_en) { + fout = DA7213_PLL_FREQ_OUT_94310400; + pll_ctrl |= DA7213_PLL_SRM_EN; + } + + /* Enable MCLK squarer if required */ + if (da7213->mclk_squarer_en) + pll_ctrl |= DA7213_PLL_MCLK_SQR_EN; + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7213_BYTE_SHIFT) & DA7213_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7213_BYTE_MASK; + + /* Write PLL dividers */ + snd_soc_write(codec, DA7213_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7213_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7213_PLL_INTEGER, pll_integer); + + /* Enable PLL */ + pll_ctrl |= DA7213_PLL_EN; + snd_soc_write(codec, DA7213_PLL_CTRL, pll_ctrl); + + return 0; + +pll_err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", + da7213->mclk_rate); + return -EINVAL; +} + +/* DAI operations */ +static const struct snd_soc_dai_ops da7213_dai_ops = { + .hw_params = da7213_hw_params, + .set_fmt = da7213_set_dai_fmt, + .set_sysclk = da7213_set_dai_sysclk, + .set_pll = da7213_set_dai_pll, + .digital_mute = da7213_mute, +}; + +static struct snd_soc_dai_driver da7213_dai = { + .name = "da7213-hifi", + /* Playback Capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7213_FORMATS, + }, + /* Capture Capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7213_FORMATS, + }, + .ops = &da7213_dai_ops, + .symmetric_rates = 1, +}; + +static int da7213_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Enable VMID reference & master bias */ + snd_soc_update_bits(codec, DA7213_REFERENCES, + DA7213_VMID_EN | DA7213_BIAS_EN, + DA7213_VMID_EN | DA7213_BIAS_EN); + } + break; + case SND_SOC_BIAS_OFF: + /* Disable VMID reference & master bias */ + snd_soc_update_bits(codec, DA7213_REFERENCES, + DA7213_VMID_EN | DA7213_BIAS_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int da7213_probe(struct snd_soc_codec *codec) +{ + int ret; + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + struct da7213_platform_data *pdata = da7213->pdata; + + codec->control_data = da7213->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Default to using ALC auto offset calibration mode. */ + snd_soc_update_bits(codec, DA7213_ALC_CTRL1, + DA7213_ALC_CALIB_MODE_MAN, 0); + da7213->alc_calib_auto = true; + + /* Default to using SRM for slave mode */ + da7213->srm_en = true; + + /* Enable all Gain Ramps */ + snd_soc_update_bits(codec, DA7213_AUX_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_AUX_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_MIXIN_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_MIXIN_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_ADC_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_ADC_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_DAC_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_DAC_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_HP_L_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_HP_R_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + snd_soc_update_bits(codec, DA7213_LINE_CTRL, + DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); + + /* + * There are two separate control bits for input and output mixers as + * well as headphone and line outs. + * One to enable corresponding amplifier and other to enable its + * output. As amplifier bits are related to power control, they are + * being managed by DAPM while other (non power related) bits are + * enabled here + */ + snd_soc_update_bits(codec, DA7213_MIXIN_L_CTRL, + DA7213_MIXIN_MIX_EN, DA7213_MIXIN_MIX_EN); + snd_soc_update_bits(codec, DA7213_MIXIN_R_CTRL, + DA7213_MIXIN_MIX_EN, DA7213_MIXIN_MIX_EN); + + snd_soc_update_bits(codec, DA7213_MIXOUT_L_CTRL, + DA7213_MIXOUT_MIX_EN, DA7213_MIXOUT_MIX_EN); + snd_soc_update_bits(codec, DA7213_MIXOUT_R_CTRL, + DA7213_MIXOUT_MIX_EN, DA7213_MIXOUT_MIX_EN); + + snd_soc_update_bits(codec, DA7213_HP_L_CTRL, + DA7213_HP_AMP_OE, DA7213_HP_AMP_OE); + snd_soc_update_bits(codec, DA7213_HP_R_CTRL, + DA7213_HP_AMP_OE, DA7213_HP_AMP_OE); + + snd_soc_update_bits(codec, DA7213_LINE_CTRL, + DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE); + + /* Set platform data values */ + if (da7213->pdata) { + u8 micbias_lvl = 0, dmic_cfg = 0; + + /* Set Mic Bias voltages */ + switch (pdata->micbias1_lvl) { + case DA7213_MICBIAS_1_6V: + case DA7213_MICBIAS_2_2V: + case DA7213_MICBIAS_2_5V: + case DA7213_MICBIAS_3_0V: + micbias_lvl |= (pdata->micbias1_lvl << + DA7213_MICBIAS1_LEVEL_SHIFT); + break; + } + switch (pdata->micbias2_lvl) { + case DA7213_MICBIAS_1_6V: + case DA7213_MICBIAS_2_2V: + case DA7213_MICBIAS_2_5V: + case DA7213_MICBIAS_3_0V: + micbias_lvl |= (pdata->micbias2_lvl << + DA7213_MICBIAS2_LEVEL_SHIFT); + break; + } + snd_soc_update_bits(codec, DA7213_MICBIAS_CTRL, + DA7213_MICBIAS1_LEVEL_MASK | + DA7213_MICBIAS2_LEVEL_MASK, micbias_lvl); + + /* Set DMIC configuration */ + switch (pdata->dmic_data_sel) { + case DA7213_DMIC_DATA_LFALL_RRISE: + case DA7213_DMIC_DATA_LRISE_RFALL: + dmic_cfg |= (pdata->dmic_data_sel << + DA7213_DMIC_DATA_SEL_SHIFT); + break; + } + switch (pdata->dmic_data_sel) { + case DA7213_DMIC_SAMPLE_ON_CLKEDGE: + case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE: + dmic_cfg |= (pdata->dmic_data_sel << + DA7213_DMIC_SAMPLEPHASE_SHIFT); + break; + } + switch (pdata->dmic_data_sel) { + case DA7213_DMIC_CLK_3_0MHZ: + case DA7213_DMIC_CLK_1_5MHZ: + dmic_cfg |= (pdata->dmic_data_sel << + DA7213_DMIC_CLK_RATE_SHIFT); + break; + } + snd_soc_update_bits(codec, DA7213_MIC_CONFIG, + DA7213_DMIC_DATA_SEL_MASK | + DA7213_DMIC_SAMPLEPHASE_MASK | + DA7213_DMIC_CLK_RATE_MASK, dmic_cfg); + + /* Set MCLK squaring */ + da7213->mclk_squarer_en = pdata->mclk_squaring; + } + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_da7213 = { + .probe = da7213_probe, + .set_bias_level = da7213_set_bias_level, + + .controls = da7213_snd_controls, + .num_controls = ARRAY_SIZE(da7213_snd_controls), + + .dapm_widgets = da7213_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7213_dapm_widgets), + .dapm_routes = da7213_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7213_audio_map), +}; + +static const struct regmap_config da7213_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da7213_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7213_reg_defaults), + .volatile_reg = da7213_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int da7213_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7213_priv *da7213; + struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev); + int ret; + + da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv), + GFP_KERNEL); + if (!da7213) + return -ENOMEM; + + if (pdata) + da7213->pdata = pdata; + + i2c_set_clientdata(i2c, da7213); + + da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config); + if (IS_ERR(da7213->regmap)) { + ret = PTR_ERR(da7213->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_da7213, &da7213_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7213 codec: %d\n", + ret); + } + return ret; +} + +static int da7213_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7213_i2c_id[] = { + { "da7213", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); + +/* I2C codec control layer */ +static struct i2c_driver da7213_i2c_driver = { + .driver = { + .name = "da7213", + .owner = THIS_MODULE, + }, + .probe = da7213_i2c_probe, + .remove = da7213_remove, + .id_table = da7213_i2c_id, +}; + +module_i2c_driver(da7213_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7213 Codec driver"); +MODULE_AUTHOR("Adam Thomson "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h new file mode 100644 index 000000000000..9cb9ddd01282 --- /dev/null +++ b/sound/soc/codecs/da7213.h @@ -0,0 +1,523 @@ +/* + * da7213.h - DA7213 ASoC Codec Driver + * + * 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_H +#define _DA7213_H + +#include +#include + +/* + * Registers + */ + +/* Status Registers */ +#define DA7213_STATUS1 0x02 +#define DA7213_PLL_STATUS 0x03 +#define DA7213_AUX_L_GAIN_STATUS 0x04 +#define DA7213_AUX_R_GAIN_STATUS 0x05 +#define DA7213_MIC_1_GAIN_STATUS 0x06 +#define DA7213_MIC_2_GAIN_STATUS 0x07 +#define DA7213_MIXIN_L_GAIN_STATUS 0x08 +#define DA7213_MIXIN_R_GAIN_STATUS 0x09 +#define DA7213_ADC_L_GAIN_STATUS 0x0A +#define DA7213_ADC_R_GAIN_STATUS 0x0B +#define DA7213_DAC_L_GAIN_STATUS 0x0C +#define DA7213_DAC_R_GAIN_STATUS 0x0D +#define DA7213_HP_L_GAIN_STATUS 0x0E +#define DA7213_HP_R_GAIN_STATUS 0x0F +#define DA7213_LINE_GAIN_STATUS 0x10 + +/* System Initialisation Registers */ +#define DA7213_DIG_ROUTING_DAI 0x21 +#define DA7213_SR 0x22 +#define DA7213_REFERENCES 0x23 +#define DA7213_PLL_FRAC_TOP 0x24 +#define DA7213_PLL_FRAC_BOT 0x25 +#define DA7213_PLL_INTEGER 0x26 +#define DA7213_PLL_CTRL 0x27 +#define DA7213_DAI_CLK_MODE 0x28 +#define DA7213_DAI_CTRL 0x29 +#define DA7213_DIG_ROUTING_DAC 0x2A +#define DA7213_ALC_CTRL1 0x2B + +/* Input - Gain, Select and Filter Registers */ +#define DA7213_AUX_L_GAIN 0x30 +#define DA7213_AUX_R_GAIN 0x31 +#define DA7213_MIXIN_L_SELECT 0x32 +#define DA7213_MIXIN_R_SELECT 0x33 +#define DA7213_MIXIN_L_GAIN 0x34 +#define DA7213_MIXIN_R_GAIN 0x35 +#define DA7213_ADC_L_GAIN 0x36 +#define DA7213_ADC_R_GAIN 0x37 +#define DA7213_ADC_FILTERS1 0x38 +#define DA7213_MIC_1_GAIN 0x39 +#define DA7213_MIC_2_GAIN 0x3A + +/* Output - Gain, Select and Filter Registers */ +#define DA7213_DAC_FILTERS5 0x40 +#define DA7213_DAC_FILTERS2 0x41 +#define DA7213_DAC_FILTERS3 0x42 +#define DA7213_DAC_FILTERS4 0x43 +#define DA7213_DAC_FILTERS1 0x44 +#define DA7213_DAC_L_GAIN 0x45 +#define DA7213_DAC_R_GAIN 0x46 +#define DA7213_CP_CTRL 0x47 +#define DA7213_HP_L_GAIN 0x48 +#define DA7213_HP_R_GAIN 0x49 +#define DA7213_LINE_GAIN 0x4A +#define DA7213_MIXOUT_L_SELECT 0x4B +#define DA7213_MIXOUT_R_SELECT 0x4C + +/* System Controller Registers */ +#define DA7213_SYSTEM_MODES_INPUT 0x50 +#define DA7213_SYSTEM_MODES_OUTPUT 0x51 + +/* Control Registers */ +#define DA7213_AUX_L_CTRL 0x60 +#define DA7213_AUX_R_CTRL 0x61 +#define DA7213_MICBIAS_CTRL 0x62 +#define DA7213_MIC_1_CTRL 0x63 +#define DA7213_MIC_2_CTRL 0x64 +#define DA7213_MIXIN_L_CTRL 0x65 +#define DA7213_MIXIN_R_CTRL 0x66 +#define DA7213_ADC_L_CTRL 0x67 +#define DA7213_ADC_R_CTRL 0x68 +#define DA7213_DAC_L_CTRL 0x69 +#define DA7213_DAC_R_CTRL 0x6A +#define DA7213_HP_L_CTRL 0x6B +#define DA7213_HP_R_CTRL 0x6C +#define DA7213_LINE_CTRL 0x6D +#define DA7213_MIXOUT_L_CTRL 0x6E +#define DA7213_MIXOUT_R_CTRL 0x6F + +/* Configuration Registers */ +#define DA7213_LDO_CTRL 0x90 +#define DA7213_IO_CTRL 0x91 +#define DA7213_GAIN_RAMP_CTRL 0x92 +#define DA7213_MIC_CONFIG 0x93 +#define DA7213_PC_COUNT 0x94 +#define DA7213_CP_VOL_THRESHOLD1 0x95 +#define DA7213_CP_DELAY 0x96 +#define DA7213_CP_DETECTOR 0x97 +#define DA7213_DAI_OFFSET 0x98 +#define DA7213_DIG_CTRL 0x99 +#define DA7213_ALC_CTRL2 0x9A +#define DA7213_ALC_CTRL3 0x9B +#define DA7213_ALC_NOISE 0x9C +#define DA7213_ALC_TARGET_MIN 0x9D +#define DA7213_ALC_TARGET_MAX 0x9E +#define DA7213_ALC_GAIN_LIMITS 0x9F +#define DA7213_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA7213_ALC_ANTICLIP_CTRL 0xA1 +#define DA7213_ALC_ANTICLIP_LEVEL 0xA2 + +#define DA7213_ALC_OFFSET_AUTO_M_L 0xA3 +#define DA7213_ALC_OFFSET_AUTO_U_L 0xA4 +#define DA7213_ALC_OFFSET_MAN_M_L 0xA6 +#define DA7213_ALC_OFFSET_MAN_U_L 0xA7 +#define DA7213_ALC_OFFSET_AUTO_M_R 0xA8 +#define DA7213_ALC_OFFSET_AUTO_U_R 0xA9 +#define DA7213_ALC_OFFSET_MAN_M_R 0xAB +#define DA7213_ALC_OFFSET_MAN_U_R 0xAC +#define DA7213_ALC_CIC_OP_LVL_CTRL 0xAD +#define DA7213_ALC_CIC_OP_LVL_DATA 0xAE +#define DA7213_DAC_NG_SETUP_TIME 0xAF +#define DA7213_DAC_NG_OFF_THRESHOLD 0xB0 +#define DA7213_DAC_NG_ON_THRESHOLD 0xB1 +#define DA7213_DAC_NG_CTRL 0xB2 + + +/* + * Bit fields + */ + +/* DA7213_SR = 0x22 */ +#define DA7213_SR_8000 (0x1 << 0) +#define DA7213_SR_11025 (0x2 << 0) +#define DA7213_SR_12000 (0x3 << 0) +#define DA7213_SR_16000 (0x5 << 0) +#define DA7213_SR_22050 (0x6 << 0) +#define DA7213_SR_24000 (0x7 << 0) +#define DA7213_SR_32000 (0x9 << 0) +#define DA7213_SR_44100 (0xA << 0) +#define DA7213_SR_48000 (0xB << 0) +#define DA7213_SR_88200 (0xE << 0) +#define DA7213_SR_96000 (0xF << 0) + +/* DA7213_REFERENCES = 0x23 */ +#define DA7213_BIAS_EN (0x1 << 3) +#define DA7213_VMID_EN (0x1 << 7) + +/* DA7213_PLL_CTRL = 0x27 */ +#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2) +#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2) +#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2) +#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2) +#define DA7213_PLL_INDIV_MASK (0x3 << 2) +#define DA7213_PLL_MCLK_SQR_EN (0x1 << 4) +#define DA7213_PLL_32K_MODE (0x1 << 5) +#define DA7213_PLL_SRM_EN (0x1 << 6) +#define DA7213_PLL_EN (0x1 << 7) + +/* DA7213_DAI_CLK_MODE = 0x28 */ +#define DA7213_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7213_DAI_CLK_POL_INV (0x1 << 2) +#define DA7213_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7) +#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7) +#define DA7213_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7213_DAI_CTRL = 0x29 */ +#define DA7213_DAI_FORMAT_I2S_MODE (0x0 << 0) +#define DA7213_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7213_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7213_DAI_FORMAT_MASK (0x3 << 0) +#define DA7213_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7213_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7213_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7213_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7213_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7213_DAI_EN_SHIFT 7 + +/* DA7213_DIG_ROUTING_DAI = 0x21 */ +#define DA7213_DAI_L_SRC_SHIFT 0 +#define DA7213_DAI_R_SRC_SHIFT 4 +#define DA7213_DAI_SRC_MAX 4 + +/* DA7213_DIG_ROUTING_DAC = 0x2A */ +#define DA7213_DAC_L_SRC_SHIFT 0 +#define DA7213_DAC_L_MONO_SHIFT 3 +#define DA7213_DAC_R_SRC_SHIFT 4 +#define DA7213_DAC_R_MONO_SHIFT 7 +#define DA7213_DAC_SRC_MAX 4 +#define DA7213_DAC_MONO_MAX 0x1 + +/* DA7213_ALC_CTRL1 = 0x2B */ +#define DA7213_ALC_OFFSET_EN_SHIFT 0 +#define DA7213_ALC_OFFSET_EN_MAX 0x1 +#define DA7213_ALC_OFFSET_EN (0x1 << 0) +#define DA7213_ALC_SYNC_MODE (0x1 << 1) +#define DA7213_ALC_CALIB_MODE_MAN (0x1 << 2) +#define DA7213_ALC_L_EN_SHIFT 3 +#define DA7213_ALC_AUTO_CALIB_EN (0x1 << 4) +#define DA7213_ALC_CALIB_OVERFLOW (0x1 << 5) +#define DA7213_ALC_R_EN_SHIFT 7 +#define DA7213_ALC_EN_MAX 0x1 + +/* DA7213_AUX_L/R_GAIN = 0x30/0x31 */ +#define DA7213_AUX_AMP_GAIN_SHIFT 0 +#define DA7213_AUX_AMP_GAIN_MAX 0x3F + +/* DA7213_MIXIN_L/R_SELECT = 0x32/0x33 */ +#define DA7213_DMIC_EN_SHIFT 7 +#define DA7213_DMIC_EN_MAX 0x1 + +/* DA7213_MIXIN_L_SELECT = 0x32 */ +#define DA7213_MIXIN_L_MIX_SELECT_AUX_L_SHIFT 0 +#define DA7213_MIXIN_L_MIX_SELECT_MIC_1_SHIFT 1 +#define DA7213_MIXIN_L_MIX_SELECT_MIC_1 (0x1 << 1) +#define DA7213_MIXIN_L_MIX_SELECT_MIC_2_SHIFT 2 +#define DA7213_MIXIN_L_MIX_SELECT_MIC_2 (0x1 << 2) +#define DA7213_MIXIN_L_MIX_SELECT_MIXIN_R_SHIFT 3 +#define DA7213_MIXIN_L_MIX_SELECT_MAX 0x1 + +/* DA7213_MIXIN_R_SELECT = 0x33 */ +#define DA7213_MIXIN_R_MIX_SELECT_AUX_R_SHIFT 0 +#define DA7213_MIXIN_R_MIX_SELECT_MIC_2_SHIFT 1 +#define DA7213_MIXIN_R_MIX_SELECT_MIC_2 (0x1 << 1) +#define DA7213_MIXIN_R_MIX_SELECT_MIC_1_SHIFT 2 +#define DA7213_MIXIN_R_MIX_SELECT_MIC_1 (0x1 << 2) +#define DA7213_MIXIN_R_MIX_SELECT_MIXIN_L_SHIFT 3 +#define DA7213_MIXIN_R_MIX_SELECT_MAX 0x1 +#define DA7213_MIC_BIAS_OUTPUT_SELECT_2 (0x1 << 6) + +/* DA7213_MIXIN_L/R_GAIN = 0x34/0x35 */ +#define DA7213_MIXIN_AMP_GAIN_SHIFT 0 +#define DA7213_MIXIN_AMP_GAIN_MAX 0xF + +/* DA7213_ADC_L/R_GAIN = 0x36/0x37 */ +#define DA7213_ADC_AMP_GAIN_SHIFT 0 +#define DA7213_ADC_AMP_GAIN_MAX 0x7F + +/* DA7213_ADC/DAC_FILTERS1 = 0x38/0x44 */ +#define DA7213_VOICE_HPF_CORNER_SHIFT 0 +#define DA7213_VOICE_HPF_CORNER_MAX 8 +#define DA7213_VOICE_EN_SHIFT 3 +#define DA7213_VOICE_EN_MAX 0x1 +#define DA7213_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7213_AUDIO_HPF_CORNER_MAX 4 +#define DA7213_HPF_EN_SHIFT 7 +#define DA7213_HPF_EN_MAX 0x1 + +/* DA7213_MIC_1/2_GAIN = 0x39/0x3A */ +#define DA7213_MIC_AMP_GAIN_SHIFT 0 +#define DA7213_MIC_AMP_GAIN_MAX 0x7 + +/* DA7213_DAC_FILTERS5 = 0x40 */ +#define DA7213_DAC_SOFTMUTE_EN_SHIFT 7 +#define DA7213_DAC_SOFTMUTE_EN_MAX 0x1 +#define DA7213_DAC_SOFTMUTE_RATE_SHIFT 4 +#define DA7213_DAC_SOFTMUTE_RATE_MAX 7 + +/* DA7213_DAC_FILTERS2/3/4 = 0x41/0x42/0x43 */ +#define DA7213_DAC_EQ_BAND_MAX 0xF + +/* DA7213_DAC_FILTERS2 = 0x41 */ +#define DA7213_DAC_EQ_BAND1_SHIFT 0 +#define DA7213_DAC_EQ_BAND2_SHIFT 4 + +/* DA7213_DAC_FILTERS2 = 0x42 */ +#define DA7213_DAC_EQ_BAND3_SHIFT 0 +#define DA7213_DAC_EQ_BAND4_SHIFT 4 + +/* DA7213_DAC_FILTERS4 = 0x43 */ +#define DA7213_DAC_EQ_BAND5_SHIFT 0 +#define DA7213_DAC_EQ_EN_SHIFT 7 +#define DA7213_DAC_EQ_EN_MAX 0x1 + +/* DA7213_DAC_L/R_GAIN = 0x45/0x46 */ +#define DA7213_DAC_AMP_GAIN_SHIFT 0 +#define DA7213_DAC_AMP_GAIN_MAX 0x7F + +/* DA7213_HP_L/R_GAIN = 0x45/0x46 */ +#define DA7213_HP_AMP_GAIN_SHIFT 0 +#define DA7213_HP_AMP_GAIN_MAX 0x3F + +/* DA7213_CP_CTRL = 0x47 */ +#define DA7213_CP_EN_SHIFT 7 + +/* DA7213_LINE_GAIN = 0x4A */ +#define DA7213_LINE_AMP_GAIN_SHIFT 0 +#define DA7213_LINE_AMP_GAIN_MAX 0x3F + +/* DA7213_MIXOUT_L_SELECT = 0x4B */ +#define DA7213_MIXOUT_L_MIX_SELECT_AUX_L_SHIFT 0 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_SHIFT 1 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_SHIFT 2 +#define DA7213_MIXOUT_L_MIX_SELECT_DAC_L_SHIFT 3 +#define DA7213_MIXOUT_L_MIX_SELECT_AUX_L_INVERTED_SHIFT 4 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_L_INVERTED_SHIFT 5 +#define DA7213_MIXOUT_L_MIX_SELECT_MIXIN_R_INVERTED_SHIFT 6 +#define DA7213_MIXOUT_L_MIX_SELECT_MAX 0x1 + +/* DA7213_MIXOUT_R_SELECT = 0x4C */ +#define DA7213_MIXOUT_R_MIX_SELECT_AUX_R_SHIFT 0 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_SHIFT 1 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_SHIFT 2 +#define DA7213_MIXOUT_R_MIX_SELECT_DAC_R_SHIFT 3 +#define DA7213_MIXOUT_R_MIX_SELECT_AUX_R_INVERTED_SHIFT 4 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_R_INVERTED_SHIFT 5 +#define DA7213_MIXOUT_R_MIX_SELECT_MIXIN_L_INVERTED_SHIFT 6 +#define DA7213_MIXOUT_R_MIX_SELECT_MAX 0x1 + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIC_1/2_CTRL = 0x63/0x64, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_ADC_L/R_CTRL = 0x65/0x66, + * DA7213_DAC_L/R_CTRL = 0x69/0x6A, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_MUTE_EN_SHIFT 6 +#define DA7213_MUTE_EN_MAX 0x1 +#define DA7213_MUTE_EN (0x1 << 6) + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_ADC_L/R_CTRL = 0x65/0x66, + * DA7213_DAC_L/R_CTRL = 0x69/0x6A, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_GAIN_RAMP_EN_SHIFT 5 +#define DA7213_GAIN_RAMP_EN_MAX 0x1 +#define DA7213_GAIN_RAMP_EN (0x1 << 5) + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_ZC_EN_SHIFT 4 +#define DA7213_ZC_EN_MAX 0x1 + +/* + * DA7213_AUX_L/R_CTRL = 0x60/0x61, + * DA7213_MIC_1/2_CTRL = 0x63/0x64, + * DA7213_MIXIN_L/R_CTRL = 0x65/0x66, + * DA7213_HP_L/R_CTRL = 0x6B/0x6C, + * DA7213_MIXOUT_L/R_CTRL = 0x6E/0x6F, + * DA7213_LINE_CTRL = 0x6D + */ +#define DA7213_AMP_EN_SHIFT 7 + +/* DA7213_MIC_1/2_CTRL = 0x63/0x64 */ +#define DA7213_MIC_AMP_IN_SEL_SHIFT 2 +#define DA7213_MIC_AMP_IN_SEL_MAX 3 + +/* DA7213_MICBIAS_CTRL = 0x62 */ +#define DA7213_MICBIAS1_LEVEL_SHIFT 0 +#define DA7213_MICBIAS1_LEVEL_MASK (0x3 << 0) +#define DA7213_MICBIAS1_EN_SHIFT 3 +#define DA7213_MICBIAS2_LEVEL_SHIFT 4 +#define DA7213_MICBIAS2_LEVEL_MASK (0x3 << 4) +#define DA7213_MICBIAS2_EN_SHIFT 7 + +/* DA7213_MIXIN_L/R_CTRL = 0x65/0x66 */ +#define DA7213_MIXIN_MIX_EN (0x1 << 3) + +/* DA7213_ADC_L/R_CTRL = 0x67/0x68 */ +#define DA7213_ADC_EN_SHIFT 7 +#define DA7213_ADC_EN (0x1 << 7) + +/* DA7213_DAC_L/R_CTRL = 0x69/0x6A*/ +#define DA7213_DAC_EN_SHIFT 7 + +/* DA7213_HP_L/R_CTRL = 0x6B/0x6C */ +#define DA7213_HP_AMP_OE (0x1 << 3) + +/* DA7213_LINE_CTRL = 0x6D */ +#define DA7213_LINE_AMP_OE (0x1 << 3) + +/* DA7213_MIXOUT_L/R_CTRL = 0x6E/0x6F */ +#define DA7213_MIXOUT_MIX_EN (0x1 << 3) + +/* DA7213_GAIN_RAMP_CTRL = 0x92 */ +#define DA7213_GAIN_RAMP_RATE_SHIFT 0 +#define DA7213_GAIN_RAMP_RATE_MAX 4 + +/* DA7213_MIC_CONFIG = 0x93 */ +#define DA7213_DMIC_DATA_SEL_SHIFT 0 +#define DA7213_DMIC_DATA_SEL_MASK (0x1 << 0) +#define DA7213_DMIC_SAMPLEPHASE_SHIFT 1 +#define DA7213_DMIC_SAMPLEPHASE_MASK (0x1 << 1) +#define DA7213_DMIC_CLK_RATE_SHIFT 2 +#define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2) + +/* DA7213_DIG_CTRL = 0x99 */ +#define DA7213_DAC_L_INV_SHIFT 3 +#define DA7213_DAC_R_INV_SHIFT 7 +#define DA7213_DAC_INV_MAX 0x1 + +/* DA7213_ALC_CTRL2 = 0x9A */ +#define DA7213_ALC_ATTACK_SHIFT 0 +#define DA7213_ALC_ATTACK_MAX 13 +#define DA7213_ALC_RELEASE_SHIFT 4 +#define DA7213_ALC_RELEASE_MAX 11 + +/* DA7213_ALC_CTRL3 = 0x9B */ +#define DA7213_ALC_HOLD_SHIFT 0 +#define DA7213_ALC_HOLD_MAX 16 +#define DA7213_ALC_INTEG_ATTACK_SHIFT 4 +#define DA7213_ALC_INTEG_RELEASE_SHIFT 6 +#define DA7213_ALC_INTEG_MAX 4 + +/* + * DA7213_ALC_NOISE = 0x9C, + * DA7213_ALC_TARGET_MIN/MAX = 0x9D/0x9E + */ +#define DA7213_ALC_THRESHOLD_SHIFT 0 +#define DA7213_ALC_THRESHOLD_MAX 0x3F + +/* DA7213_ALC_GAIN_LIMITS = 0x9F */ +#define DA7213_ALC_ATTEN_MAX_SHIFT 0 +#define DA7213_ALC_GAIN_MAX_SHIFT 4 +#define DA7213_ALC_ATTEN_GAIN_MAX_MAX 0xF + +/* DA7213_ALC_ANA_GAIN_LIMITS = 0xA0 */ +#define DA7213_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7213_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7213_ALC_ANA_GAIN_MAX 0x7 + +/* DA7213_ALC_ANTICLIP_CTRL = 0xA1 */ +#define DA7213_ALC_ANTICLIP_EN_SHIFT 7 +#define DA7213_ALC_ANTICLIP_EN_MAX 0x1 + +/* DA7213_ALC_ANTICLIP_LEVEL = 0xA2 */ +#define DA7213_ALC_ANTICLIP_LEVEL_SHIFT 0 +#define DA7213_ALC_ANTICLIP_LEVEL_MAX 0x7F + +/* DA7213_ALC_CIC_OP_LVL_CTRL = 0xAD */ +#define DA7213_ALC_DATA_MIDDLE (0x2 << 0) +#define DA7213_ALC_DATA_TOP (0x3 << 0) +#define DA7213_ALC_CIC_OP_CHANNEL_LEFT (0x0 << 7) +#define DA7213_ALC_CIC_OP_CHANNEL_RIGHT (0x1 << 7) + +/* DA7213_DAC_NG_SETUP_TIME = 0xAF */ +#define DA7213_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7213_DAC_NG_SETUP_TIME_MAX 4 +#define DA7213_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7213_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7213_DAC_NG_RAMP_RATE_MAX 2 + +/* DA7213_DAC_NG_OFF/ON_THRESH = 0xB0/0xB1 */ +#define DA7213_DAC_NG_THRESHOLD_SHIFT 0 +#define DA7213_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7213_DAC_NG_CTRL = 0xB2 */ +#define DA7213_DAC_NG_EN_SHIFT 7 +#define DA7213_DAC_NG_EN_MAX 0x1 + + +/* + * General defines + */ + +/* Register inversion */ +#define DA7213_NO_INVERT 0 +#define DA7213_INVERT 1 + +/* Byte related defines */ +#define DA7213_BYTE_SHIFT 8 +#define DA7213_BYTE_MASK 0xFF + +/* ALC related */ +#define DA7213_ALC_OFFSET_15_8 0x00FF00 +#define DA7213_ALC_OFFSET_19_16 0x0F0000 +#define DA7213_ALC_AVG_ITERATIONS 5 + +/* PLL related */ +#define DA7213_SYSCLK_MCLK 0 +#define DA7213_SYSCLK_PLL 1 +#define DA7213_PLL_FREQ_OUT_90316800 90316800 +#define DA7213_PLL_FREQ_OUT_98304000 98304000 +#define DA7213_PLL_FREQ_OUT_94310400 94310400 +#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2 +#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4 +#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 +#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 + +enum clk_src { + DA7213_CLKSRC_MCLK +}; + +/* Codec private data */ +struct da7213_priv { + struct regmap *regmap; + unsigned int mclk_rate; + bool master; + bool mclk_squarer_en; + bool srm_en; + bool alc_calib_auto; + bool alc_en; + struct da7213_platform_data *pdata; +}; + +#endif /* _DA7213_H */ -- cgit v1.2.3 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 ac90037894d9..d2314be4f0c0 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