From 8ecb5344fd6409c500c9d5757c3a7130d3d7db5b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:51:38 +0100 Subject: ASoC: cq93vc: Don't use control data for core driver data The platform data is being used to obtain the core driver data for the device (which is a bit of an abuse but not the issue at hand) so reference it directly in order to support refactoring to use regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 23316c887b19..e2c4c0a896e2 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -79,7 +79,7 @@ static int cq93vc_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 davinci_vc *davinci_vc = codec->control_data; + struct davinci_vc *davinci_vc = codec->dev->platform_data; switch (freq) { case 22579200: -- cgit From 89d051300b845e1001a9d9e9ce94da4250c21613 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:22:18 +0530 Subject: ASoC: rt5640: Staticize hp_amp_power_on 'hp_amp_power_on' is used only in this file. Make it static. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c26a8f814b18..2f6bb161e64c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -926,7 +926,7 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } -void hp_amp_power_on(struct snd_soc_codec *codec) +static void hp_amp_power_on(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); -- cgit From 9e9cb9b99615180b94d743f1d6ca0f82539c8754 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 17:57:35 +0100 Subject: ASoC: rt5640: Provide more useful hw_params error reasons. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 2f6bb161e64c..de40bd9f6ac2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1609,7 +1609,8 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5640->lrck[dai->id], dai->id); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- cgit From 02b80773de3732dae11c1cf0c1ce40378901bd0e Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 13 Sep 2013 17:57:36 +0100 Subject: ASoC: rt5640: Add ACPI probing support. Allow the RT5640 to be probed as an ACPI I2C device. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de40bd9f6ac2..0bfb960e90f8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -2081,6 +2082,12 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static struct acpi_device_id rt5640_acpi_match[] = { + { "INT33CA", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); + static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { rt5640->pdata.in1_diff = of_property_read_bool(np, @@ -2200,6 +2207,7 @@ static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5640_acpi_match), }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, -- cgit From 37c83edf9afd3d7b39ace9113a166c03b7a2820f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:17:08 +0100 Subject: ASoC: wm8400: Use supplies to manage input power Rather than using a fake register to manage input power create some supply widgets and use those. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 73 ++++++++++++----------------------------------- 1 file changed, 18 insertions(+), 55 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index d2a092850283..95c33d169952 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -32,13 +32,6 @@ #include "wm8400.h" -/* Fake register for internal state */ -#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) -#define WM8400_INMIXL_PWR 0 -#define WM8400_AINLMUX_PWR 1 -#define WM8400_INMIXR_PWR 2 -#define WM8400_AINRMUX_PWR 3 - static struct regulator_bulk_data power[] = { { .supply = "I2S1VDD", @@ -79,10 +72,7 @@ static inline unsigned int wm8400_read(struct snd_soc_codec *codec, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) - return wm8400->fake_register; - else - return wm8400_reg_read(wm8400->wm8400, reg); + return wm8400_reg_read(wm8400->wm8400, reg); } /* @@ -93,11 +83,7 @@ static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) { - wm8400->fake_register = value; - return 0; - } else - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); } static void wm8400_codec_reset(struct snd_soc_codec *codec) @@ -352,32 +338,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event (struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); - - if (fakepower & ((1 << WM8400_INMIXL_PWR) | - (1 << WM8400_AINLMUX_PWR))) { - reg |= WM8400_AINL_ENA; - } else { - reg &= ~WM8400_AINL_ENA; - } - - if (fakepower & ((1 << WM8400_INMIXR_PWR) | - (1 << WM8400_AINRMUX_PWR))) { - reg |= WM8400_AINR_ENA; - } else { - reg &= ~WM8400_AINR_ENA; - } - snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event (struct snd_soc_dapm_widget *w, struct snd_kcontrol * kcontrol, int event) { @@ -658,27 +618,26 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, 0, &wm8400_dapm_rin34_pga_controls[0], ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), +SND_SOC_DAPM_SUPPLY("INL", WM8400_POWER_MANAGEMENT_2, WM8400_AINL_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8400_POWER_MANAGEMENT_2, WM8400_AINR_ENA_SHIFT, + 0, NULL, 0), + /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, - &wm8400_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AILNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, - &wm8400_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AIRNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -789,11 +748,13 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { {"LIN34 PGA", "LIN3 Switch", "LIN3"}, {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ + {"INMIXL", NULL, "INL"}, {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, /* AILNMUX */ + {"AILNMUX", NULL, "INL"}, {"AILNMUX", "INMIXL Mix", "INMIXL"}, {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, @@ -808,12 +769,14 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, - /* INMIXL */ + /* INMIXR */ + {"INMIXR", NULL, "INR"}, {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, /* AIRNMUX */ + {"AIRNMUX", NULL, "INR"}, {"AIRNMUX", "INMIXR Mix", "INMIXR"}, {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, -- cgit From b8cc4151f8af97e1b573ca399a77f439f401a57e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:21:12 +0100 Subject: ASoC: wm8400: Use regmap for I/O Since we no longer have a fake register to simulate we can use the framework for I/O. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 95c33d169952..48dc7d2fee36 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -67,25 +67,6 @@ struct wm8400_priv { int fll_in, fll_out; }; -static inline unsigned int wm8400_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - - return wm8400_reg_read(wm8400->wm8400, reg); -} - -/* - * write to the wm8400 register space - */ -static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); -} - static void wm8400_codec_reset(struct snd_soc_codec *codec) { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); @@ -1328,9 +1309,12 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); - codec->control_data = priv->wm8400 = wm8400; + priv->wm8400 = wm8400; + codec->control_data = wm8400->regmap; priv->codec = codec; + snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); if (ret != 0) { @@ -1377,8 +1361,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .remove = wm8400_codec_remove, .suspend = wm8400_suspend, .resume = wm8400_resume, - .read = snd_soc_read, - .write = wm8400_write, .set_bias_level = wm8400_set_bias_level, .controls = wm8400_snd_controls, -- cgit From 49c60547daebaa79e8de9d2dff6dee994576c94c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Sep 2013 15:34:35 +0100 Subject: ASoC: arizona: Improve handling of setting REFCLK to 0 This patch suppresses calculation of REFCLK parameters when the REFCLK source frequency is set to zero, additionally it will consider a source frequency of zero as the REFCLK being disabled and switch to using the SYNCCLK. Reported-by: Kyung Kwee Ryu Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808ba1418..6f05b17d1965 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll, { struct arizona *arizona = fll->arizona; int ret; + bool use_sync = false; /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. */ - if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + if (fll->ref_src >= 0 && fll->ref_freq && + fll->ref_src != fll->sync_src) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); - if (fll->sync_src >= 0) + if (fll->sync_src >= 0) { arizona_apply_fll(arizona, fll->base + 0x10, sync, fll->sync_src, true); + use_sync = true; + } } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, @@ -1511,7 +1515,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, * Increase the bandwidth if we're not using a low frequency * sync source. */ - if (fll->sync_src >= 0 && fll->sync_freq > 100000) + if (use_sync && fll->sync_freq > 100000) regmap_update_bits(arizona->regmap, fll->base + 0x17, ARIZONA_FLL1_SYNC_BW, 0); else @@ -1526,8 +1530,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->ref_src >= 0 && fll->sync_src >= 0 && - fll->ref_src != fll->sync_src) + if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1561,10 +1564,12 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout && Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; + if (fll->fout) { + if (Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + } if (fll->sync_src >= 0) { ret = arizona_calc_fll(fll, &sync, fll->sync_freq, -- cgit From 193b2f65b87e9da78b15f3e3a0cae1d37fbafa57 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 16 Sep 2013 18:14:20 +0200 Subject: ASoC: ak4104: provide a module device table Provide a module device table for the SPI subsystem, so the driver can be autoloaded by the SPI core. While at it, get rid of an unnecessary #define. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 71059c07ae7b..b4819dcd4f4d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -45,8 +45,6 @@ #define AK4104_TX_TXE (1 << 0) #define AK4104_TX_V (1 << 1) -#define DRV_NAME "ak4104-codec" - struct ak4104_private { struct regmap *regmap; }; @@ -291,12 +289,19 @@ static const struct of_device_id ak4104_of_match[] = { }; MODULE_DEVICE_TABLE(of, ak4104_of_match); +static const struct spi_device_id ak4104_id_table[] = { + { "ak4104", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, ak4104_id_table); + static struct spi_driver ak4104_spi_driver = { .driver = { - .name = DRV_NAME, + .name = "ak4104", .owner = THIS_MODULE, .of_match_table = ak4104_of_match, }, + .id_table = ak4104_id_table, .probe = ak4104_spi_probe, .remove = ak4104_spi_remove, }; -- cgit From bf551413038f74343ec4d1413c3610e2362d0aeb Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 16:16:17 +0530 Subject: ASoC: twl6040: Remove redundant semicolon Redundant semicolon removed. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb6c323..35059a242fa4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -246,7 +246,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -1100,7 +1100,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) -- cgit From 0feb23d1bdf31db903069d3d94892e56b5c11981 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 13 Sep 2013 15:50:50 +0530 Subject: ASoC: ak4642: Remove redundant break 'break' after return statement is redundant. Remove it. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d0378709702..21c35ed778cc 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); @@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); -- cgit From 32fcb97b9f699f63742bcaadca6e0beede86e8e8 Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Thu, 19 Sep 2013 11:18:06 +0200 Subject: ASoC: rt5640: Omit ACPI match table only if !ACPI The ACPI_PTR() macro evaluates to NULL if ACPI is disabled and hence the ACPI match table won't be used, causing the compiler to complain. Avoid this by protecting the table using an #ifdef CONFIG_ACPI. Signed-off-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0bfb960e90f8..641eeeb00c5c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2082,11 +2082,13 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); +#endif static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { -- cgit From fa129ebeba6db2b4bcea45efe87a71d68181c04c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 18:20:26 +0100 Subject: ASoC: 88pm60x: Don't use control data for i2c In preparation for using the regmap directly in the CODEC driver replace references to the I2C client using control_data with references to the driver private data. Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 8af04343cc1a..3925cf34f751 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1166,6 +1166,7 @@ static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, static int pm860x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); int data; switch (level) { @@ -1179,17 +1180,17 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); } break; case SND_SOC_BIAS_OFF: data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); + pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0); break; } codec->dapm.bias_level = level; @@ -1319,17 +1320,17 @@ int pm860x_hs_jack_detect(struct snd_soc_codec *codec, pm860x->det.lo_shrt = lo_shrt; if (det & SND_JACK_HEADPHONE) - pm860x_set_bits(codec->control_data, REG_HS_DET, + pm860x_set_bits(pm860x->i2c, REG_HS_DET, EN_HS_DET, EN_HS_DET); /* headset short detect */ if (hs_shrt) { data = CLR_SHORT_HS2 | CLR_SHORT_HS1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* Lineout short detect */ if (lo_shrt) { data = CLR_SHORT_LO2 | CLR_SHORT_LO1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* sync status */ @@ -1347,7 +1348,7 @@ int pm860x_mic_jack_detect(struct snd_soc_codec *codec, pm860x->det.mic_det = det; if (det & SND_JACK_MICROPHONE) - pm860x_set_bits(codec->control_data, REG_MIC_DET, + pm860x_set_bits(pm860x->i2c, REG_MIC_DET, MICDET_MASK, MICDET_MASK); /* sync status */ @@ -1377,7 +1378,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE, + ret = pm860x_bulk_read(pm860x->i2c, REG_CACHE_BASE, REG_CACHE_SIZE, codec->reg_cache); if (ret < 0) { dev_err(codec->dev, "Failed to fill register cache: %d\n", -- cgit From f9ded3b2e761256301ebb8d90e87eb1b5443e3ea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 19:00:46 +0100 Subject: ASoC: 88pm860x: Use regmap for I/O As part of a move to remove the duplication of regmap functionality in ASoC convert the 88pm860x driver to use the regmap from the MFD. This means that we no longer cache the registers so performance will be slightly reduced on I/O operations. Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 60 ++----------------- sound/soc/codecs/88pm860x-codec.h | 117 +++++++++++++++++++------------------- 2 files changed, 63 insertions(+), 114 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 3925cf34f751..4633e51b1500 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -140,6 +141,7 @@ struct pm860x_priv { unsigned int filter; struct snd_soc_codec *codec; struct i2c_client *i2c; + struct regmap *regmap; struct pm860x_chip *chip; struct pm860x_det det; @@ -269,48 +271,6 @@ static struct st_gain st_table[] = { { -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0}, }; -static int pm860x_volatile(unsigned int reg) -{ - BUG_ON(reg >= REG_CACHE_SIZE); - - switch (reg) { - case PM860X_AUDIO_SUPPLIES_2: - return 1; - } - - return 0; -} - -static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned char *cache = codec->reg_cache; - - BUG_ON(reg >= REG_CACHE_SIZE); - - if (pm860x_volatile(reg)) - return cache[reg]; - - reg += REG_CACHE_BASE; - - return pm860x_reg_read(codec->control_data, reg); -} - -static int pm860x_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - unsigned char *cache = codec->reg_cache; - - BUG_ON(reg >= REG_CACHE_SIZE); - - if (!pm860x_volatile(reg)) - cache[reg] = (unsigned char)value; - - reg += REG_CACHE_BASE; - - return pm860x_reg_write(codec->control_data, reg, value); -} - static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1364,7 +1324,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; - codec->control_data = pm860x->i2c; + codec->control_data = pm860x->regmap; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, @@ -1378,14 +1338,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(pm860x->i2c, REG_CACHE_BASE, - REG_CACHE_SIZE, codec->reg_cache); - if (ret < 0) { - dev_err(codec->dev, "Failed to fill register cache: %d\n", - ret); - goto out; - } - return 0; out: @@ -1408,10 +1360,6 @@ static int pm860x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pm860x = { .probe = pm860x_probe, .remove = pm860x_remove, - .read = pm860x_read_reg_cache, - .write = pm860x_write_reg_cache, - .reg_cache_size = REG_CACHE_SIZE, - .reg_word_size = sizeof(u8), .set_bias_level = pm860x_set_bias_level, .controls = pm860x_snd_controls, @@ -1437,6 +1385,8 @@ static int pm860x_codec_probe(struct platform_device *pdev) pm860x->chip = chip; pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client : chip->companion; + pm860x->regmap = (chip->id == CHIP_PM8607) ? chip->regmap + : chip->regmap_companion; platform_set_drvdata(pdev, pm860x); for (i = 0; i < 4; i++) { diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h index 3364ba4a3607..f7282f4f4a79 100644 --- a/sound/soc/codecs/88pm860x-codec.h +++ b/sound/soc/codecs/88pm860x-codec.h @@ -12,67 +12,66 @@ #ifndef __88PM860X_H #define __88PM860X_H -/* The offset of these registers are 0xb0 */ -#define PM860X_PCM_IFACE_1 0x00 -#define PM860X_PCM_IFACE_2 0x01 -#define PM860X_PCM_IFACE_3 0x02 -#define PM860X_PCM_RATE 0x03 -#define PM860X_EC_PATH 0x04 -#define PM860X_SIDETONE_L_GAIN 0x05 -#define PM860X_SIDETONE_R_GAIN 0x06 -#define PM860X_SIDETONE_SHIFT 0x07 -#define PM860X_ADC_OFFSET_1 0x08 -#define PM860X_ADC_OFFSET_2 0x09 -#define PM860X_DMIC_DELAY 0x0a +#define PM860X_PCM_IFACE_1 0xb0 +#define PM860X_PCM_IFACE_2 0xb1 +#define PM860X_PCM_IFACE_3 0xb2 +#define PM860X_PCM_RATE 0xb3 +#define PM860X_EC_PATH 0xb4 +#define PM860X_SIDETONE_L_GAIN 0xb5 +#define PM860X_SIDETONE_R_GAIN 0xb6 +#define PM860X_SIDETONE_SHIFT 0xb7 +#define PM860X_ADC_OFFSET_1 0xb8 +#define PM860X_ADC_OFFSET_2 0xb9 +#define PM860X_DMIC_DELAY 0xba -#define PM860X_I2S_IFACE_1 0x0b -#define PM860X_I2S_IFACE_2 0x0c -#define PM860X_I2S_IFACE_3 0x0d -#define PM860X_I2S_IFACE_4 0x0e -#define PM860X_EQUALIZER_N0_1 0x0f -#define PM860X_EQUALIZER_N0_2 0x10 -#define PM860X_EQUALIZER_N1_1 0x11 -#define PM860X_EQUALIZER_N1_2 0x12 -#define PM860X_EQUALIZER_D1_1 0x13 -#define PM860X_EQUALIZER_D1_2 0x14 -#define PM860X_LOFI_GAIN_LEFT 0x15 -#define PM860X_LOFI_GAIN_RIGHT 0x16 -#define PM860X_HIFIL_GAIN_LEFT 0x17 -#define PM860X_HIFIL_GAIN_RIGHT 0x18 -#define PM860X_HIFIR_GAIN_LEFT 0x19 -#define PM860X_HIFIR_GAIN_RIGHT 0x1a -#define PM860X_DAC_OFFSET 0x1b -#define PM860X_OFFSET_LEFT_1 0x1c -#define PM860X_OFFSET_LEFT_2 0x1d -#define PM860X_OFFSET_RIGHT_1 0x1e -#define PM860X_OFFSET_RIGHT_2 0x1f -#define PM860X_ADC_ANA_1 0x20 -#define PM860X_ADC_ANA_2 0x21 -#define PM860X_ADC_ANA_3 0x22 -#define PM860X_ADC_ANA_4 0x23 -#define PM860X_ANA_TO_ANA 0x24 -#define PM860X_HS1_CTRL 0x25 -#define PM860X_HS2_CTRL 0x26 -#define PM860X_LO1_CTRL 0x27 -#define PM860X_LO2_CTRL 0x28 -#define PM860X_EAR_CTRL_1 0x29 -#define PM860X_EAR_CTRL_2 0x2a -#define PM860X_AUDIO_SUPPLIES_1 0x2b -#define PM860X_AUDIO_SUPPLIES_2 0x2c -#define PM860X_ADC_EN_1 0x2d -#define PM860X_ADC_EN_2 0x2e -#define PM860X_DAC_EN_1 0x2f -#define PM860X_DAC_EN_2 0x31 -#define PM860X_AUDIO_CAL_1 0x32 -#define PM860X_AUDIO_CAL_2 0x33 -#define PM860X_AUDIO_CAL_3 0x34 -#define PM860X_AUDIO_CAL_4 0x35 -#define PM860X_AUDIO_CAL_5 0x36 -#define PM860X_ANA_INPUT_SEL_1 0x37 -#define PM860X_ANA_INPUT_SEL_2 0x38 +#define PM860X_I2S_IFACE_1 0xbb +#define PM860X_I2S_IFACE_2 0xbc +#define PM860X_I2S_IFACE_3 0xbd +#define PM860X_I2S_IFACE_4 0xbe +#define PM860X_EQUALIZER_N0_1 0xbf +#define PM860X_EQUALIZER_N0_2 0xc0 +#define PM860X_EQUALIZER_N1_1 0xc1 +#define PM860X_EQUALIZER_N1_2 0xc2 +#define PM860X_EQUALIZER_D1_1 0xc3 +#define PM860X_EQUALIZER_D1_2 0xc4 +#define PM860X_LOFI_GAIN_LEFT 0xc5 +#define PM860X_LOFI_GAIN_RIGHT 0xc6 +#define PM860X_HIFIL_GAIN_LEFT 0xc7 +#define PM860X_HIFIL_GAIN_RIGHT 0xc8 +#define PM860X_HIFIR_GAIN_LEFT 0xc9 +#define PM860X_HIFIR_GAIN_RIGHT 0xca +#define PM860X_DAC_OFFSET 0xcb +#define PM860X_OFFSET_LEFT_1 0xcc +#define PM860X_OFFSET_LEFT_2 0xcd +#define PM860X_OFFSET_RIGHT_1 0xce +#define PM860X_OFFSET_RIGHT_2 0xcf +#define PM860X_ADC_ANA_1 0xd0 +#define PM860X_ADC_ANA_2 0xd1 +#define PM860X_ADC_ANA_3 0xd2 +#define PM860X_ADC_ANA_4 0xd3 +#define PM860X_ANA_TO_ANA 0xd4 +#define PM860X_HS1_CTRL 0xd5 +#define PM860X_HS2_CTRL 0xd6 +#define PM860X_LO1_CTRL 0xd7 +#define PM860X_LO2_CTRL 0xd8 +#define PM860X_EAR_CTRL_1 0xd9 +#define PM860X_EAR_CTRL_2 0xda +#define PM860X_AUDIO_SUPPLIES_1 0xdb +#define PM860X_AUDIO_SUPPLIES_2 0xdc +#define PM860X_ADC_EN_1 0xdd +#define PM860X_ADC_EN_2 0xde +#define PM860X_DAC_EN_1 0xdf +#define PM860X_DAC_EN_2 0xe1 +#define PM860X_AUDIO_CAL_1 0xe2 +#define PM860X_AUDIO_CAL_2 0xe3 +#define PM860X_AUDIO_CAL_3 0xe4 +#define PM860X_AUDIO_CAL_4 0xe5 +#define PM860X_AUDIO_CAL_5 0xe6 +#define PM860X_ANA_INPUT_SEL_1 0xe7 +#define PM860X_ANA_INPUT_SEL_2 0xe8 -#define PM860X_PCM_IFACE_4 0x39 -#define PM860X_I2S_IFACE_5 0x3a +#define PM860X_PCM_IFACE_4 0xe9 +#define PM860X_I2S_IFACE_5 0xea #define PM860X_SHORTS 0x3b #define PM860X_PLL_ADJ_1 0x3c -- cgit From 38bfd48b87c44f6958f75bfcd5ae5a53bd3ca07b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 19:17:14 +0100 Subject: ASoC: ab8500: Downgrade noisy log message Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0adacfce..7cea5a8487d0 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2601,7 +2601,7 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) static int ab8500_codec_driver_remove(struct platform_device *pdev) { - dev_info(&pdev->dev, "%s Enter.\n", __func__); + dev_dbg(&pdev->dev, "%s Enter.\n", __func__); snd_soc_unregister_codec(&pdev->dev); -- cgit From 51f20e4cd83e804fb4fd940873763f29616f12a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Sep 2013 19:25:18 +0100 Subject: ASoC: ab8500: Use ASoC I/O functions In preparation for moving away from implementing the ASoC level register I/O functionality change direct calls to the ab8500 implementation of that to use snd_soc_write() Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 7cea5a8487d0..c2b663696611 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2527,12 +2527,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) } /* Override HW-defaults */ - ab8500_codec_write_reg(codec, - AB8500_ANACONF5, - BIT(AB8500_ANACONF5_HSAUTOEN)); - ab8500_codec_write_reg(codec, - AB8500_SHORTCIRCONF, - BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); + snd_soc_write(codec, AB8500_ANACONF5, + BIT(AB8500_ANACONF5_HSAUTOEN)); + snd_soc_write(codec, AB8500_SHORTCIRCONF, + BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); /* Add filter controls */ status = snd_soc_add_codec_controls(codec, ab8500_filter_controls, -- cgit From ff795d614bfa62a3c6fc0bcb75cb8842e5a87892 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 20 Sep 2013 10:38:16 +0100 Subject: ASoC: ab8500: Convert register I/O to regmap As part of a general push to eliminate the duplicated register I/O support in ASoC convert ab8500 to use regmap. Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 64 +++++++++++++++++++---------------------- 1 file changed, 30 insertions(+), 34 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index c2b663696611..d5a0fc4b2fe2 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -126,6 +126,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -166,49 +168,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; + struct device *dev = context; - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); - - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -2483,6 +2471,8 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); + codec->control_data = drvdata->regmap; + if (np) { if (!pdata) pdata = devm_kzalloc(dev, @@ -2560,9 +2550,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2585,6 +2572,15 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, -- cgit From 356d86e24850cdc353602b90be73e627f86707c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 17:22:17 +0100 Subject: ASoC: max98088: Fix indentation Tested-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 46 ++++++++++++++++++++++----------------------- 1 file changed, 23 insertions(+), 23 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 566a367c94fa..391f66913a44 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -38,20 +38,20 @@ struct max98088_cdata { }; struct max98088_priv { - enum max98088_type devtype; - struct max98088_pdata *pdata; - unsigned int sysclk; - struct max98088_cdata dai[2]; - int eq_textcnt; - const char **eq_texts; - struct soc_enum eq_enum; - u8 ina_state; - u8 inb_state; - unsigned int ex_mode; - unsigned int digmic; - unsigned int mic1pre; - unsigned int mic2pre; - unsigned int extmic_mode; + enum max98088_type devtype; + struct max98088_pdata *pdata; + unsigned int sysclk; + struct max98088_cdata dai[2]; + int eq_textcnt; + const char **eq_texts; + struct soc_enum eq_enum; + u8 ina_state; + u8 inb_state; + unsigned int ex_mode; + unsigned int digmic; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mode; }; static const u8 max98088_reg[M98088_REG_CNT] = { @@ -2066,15 +2066,15 @@ static int max98088_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_max98088 = { - .probe = max98088_probe, - .remove = max98088_remove, - .suspend = max98088_suspend, - .resume = max98088_resume, - .set_bias_level = max98088_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, + .probe = max98088_probe, + .remove = max98088_remove, + .suspend = max98088_suspend, + .resume = max98088_resume, + .set_bias_level = max98088_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max98088_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max98088_reg, + .volatile_register = max98088_volatile_register, .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, -- cgit From ad65adf4a3039ecd93d4712ac6524dbd9e0e848a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 17:54:02 +0100 Subject: ASoC: max98088: Use table based control init Tested-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 391f66913a44..8896d5e33980 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2048,9 +2048,6 @@ static int max98088_probe(struct snd_soc_codec *codec) max98088_handle_pdata(codec); - snd_soc_add_codec_controls(codec, max98088_snd_controls, - ARRAY_SIZE(max98088_snd_controls)); - err_access: return ret; } @@ -2071,6 +2068,8 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { .suspend = max98088_suspend, .resume = max98088_resume, .set_bias_level = max98088_set_bias_level, + .controls = max98088_snd_controls, + .num_controls = ARRAY_SIZE(max98088_snd_controls), .reg_cache_size = ARRAY_SIZE(max98088_reg), .reg_word_size = sizeof(u8), .reg_cache_default = max98088_reg, -- cgit From 4127d5d59f8135e3c187b8daa2540691761938ce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 17:56:17 +0100 Subject: ASoC: max98088: Convert to direct regmap API usage This saves code and moves us towards removing the redundant register I/O implementation in ASoC. Tested-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 580 +++++++++++++++++++------------------------- 1 file changed, 251 insertions(+), 329 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 8896d5e33980..31912d59702c 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -38,6 +39,7 @@ struct max98088_cdata { }; struct max98088_priv { + struct regmap *regmap; enum max98088_type devtype; struct max98088_pdata *pdata; unsigned int sysclk; @@ -54,278 +56,206 @@ struct max98088_priv { unsigned int extmic_mode; }; -static const u8 max98088_reg[M98088_REG_CNT] = { - 0x00, /* 00 IRQ status */ - 0x00, /* 01 MIC status */ - 0x00, /* 02 jack status */ - 0x00, /* 03 battery voltage */ - 0x00, /* 04 */ - 0x00, /* 05 */ - 0x00, /* 06 */ - 0x00, /* 07 */ - 0x00, /* 08 */ - 0x00, /* 09 */ - 0x00, /* 0A */ - 0x00, /* 0B */ - 0x00, /* 0C */ - 0x00, /* 0D */ - 0x00, /* 0E */ - 0x00, /* 0F interrupt enable */ - - 0x00, /* 10 master clock */ - 0x00, /* 11 DAI1 clock mode */ - 0x00, /* 12 DAI1 clock control */ - 0x00, /* 13 DAI1 clock control */ - 0x00, /* 14 DAI1 format */ - 0x00, /* 15 DAI1 clock */ - 0x00, /* 16 DAI1 config */ - 0x00, /* 17 DAI1 TDM */ - 0x00, /* 18 DAI1 filters */ - 0x00, /* 19 DAI2 clock mode */ - 0x00, /* 1A DAI2 clock control */ - 0x00, /* 1B DAI2 clock control */ - 0x00, /* 1C DAI2 format */ - 0x00, /* 1D DAI2 clock */ - 0x00, /* 1E DAI2 config */ - 0x00, /* 1F DAI2 TDM */ - - 0x00, /* 20 DAI2 filters */ - 0x00, /* 21 data config */ - 0x00, /* 22 DAC mixer */ - 0x00, /* 23 left ADC mixer */ - 0x00, /* 24 right ADC mixer */ - 0x00, /* 25 left HP mixer */ - 0x00, /* 26 right HP mixer */ - 0x00, /* 27 HP control */ - 0x00, /* 28 left REC mixer */ - 0x00, /* 29 right REC mixer */ - 0x00, /* 2A REC control */ - 0x00, /* 2B left SPK mixer */ - 0x00, /* 2C right SPK mixer */ - 0x00, /* 2D SPK control */ - 0x00, /* 2E sidetone */ - 0x00, /* 2F DAI1 playback level */ - - 0x00, /* 30 DAI1 playback level */ - 0x00, /* 31 DAI2 playback level */ - 0x00, /* 32 DAI2 playbakc level */ - 0x00, /* 33 left ADC level */ - 0x00, /* 34 right ADC level */ - 0x00, /* 35 MIC1 level */ - 0x00, /* 36 MIC2 level */ - 0x00, /* 37 INA level */ - 0x00, /* 38 INB level */ - 0x00, /* 39 left HP volume */ - 0x00, /* 3A right HP volume */ - 0x00, /* 3B left REC volume */ - 0x00, /* 3C right REC volume */ - 0x00, /* 3D left SPK volume */ - 0x00, /* 3E right SPK volume */ - 0x00, /* 3F MIC config */ - - 0x00, /* 40 MIC threshold */ - 0x00, /* 41 excursion limiter filter */ - 0x00, /* 42 excursion limiter threshold */ - 0x00, /* 43 ALC */ - 0x00, /* 44 power limiter threshold */ - 0x00, /* 45 power limiter config */ - 0x00, /* 46 distortion limiter config */ - 0x00, /* 47 audio input */ - 0x00, /* 48 microphone */ - 0x00, /* 49 level control */ - 0x00, /* 4A bypass switches */ - 0x00, /* 4B jack detect */ - 0x00, /* 4C input enable */ - 0x00, /* 4D output enable */ - 0xF0, /* 4E bias control */ - 0x00, /* 4F DAC power */ - - 0x0F, /* 50 DAC power */ - 0x00, /* 51 system */ - 0x00, /* 52 DAI1 EQ1 */ - 0x00, /* 53 DAI1 EQ1 */ - 0x00, /* 54 DAI1 EQ1 */ - 0x00, /* 55 DAI1 EQ1 */ - 0x00, /* 56 DAI1 EQ1 */ - 0x00, /* 57 DAI1 EQ1 */ - 0x00, /* 58 DAI1 EQ1 */ - 0x00, /* 59 DAI1 EQ1 */ - 0x00, /* 5A DAI1 EQ1 */ - 0x00, /* 5B DAI1 EQ1 */ - 0x00, /* 5C DAI1 EQ2 */ - 0x00, /* 5D DAI1 EQ2 */ - 0x00, /* 5E DAI1 EQ2 */ - 0x00, /* 5F DAI1 EQ2 */ - - 0x00, /* 60 DAI1 EQ2 */ - 0x00, /* 61 DAI1 EQ2 */ - 0x00, /* 62 DAI1 EQ2 */ - 0x00, /* 63 DAI1 EQ2 */ - 0x00, /* 64 DAI1 EQ2 */ - 0x00, /* 65 DAI1 EQ2 */ - 0x00, /* 66 DAI1 EQ3 */ - 0x00, /* 67 DAI1 EQ3 */ - 0x00, /* 68 DAI1 EQ3 */ - 0x00, /* 69 DAI1 EQ3 */ - 0x00, /* 6A DAI1 EQ3 */ - 0x00, /* 6B DAI1 EQ3 */ - 0x00, /* 6C DAI1 EQ3 */ - 0x00, /* 6D DAI1 EQ3 */ - 0x00, /* 6E DAI1 EQ3 */ - 0x00, /* 6F DAI1 EQ3 */ - - 0x00, /* 70 DAI1 EQ4 */ - 0x00, /* 71 DAI1 EQ4 */ - 0x00, /* 72 DAI1 EQ4 */ - 0x00, /* 73 DAI1 EQ4 */ - 0x00, /* 74 DAI1 EQ4 */ - 0x00, /* 75 DAI1 EQ4 */ - 0x00, /* 76 DAI1 EQ4 */ - 0x00, /* 77 DAI1 EQ4 */ - 0x00, /* 78 DAI1 EQ4 */ - 0x00, /* 79 DAI1 EQ4 */ - 0x00, /* 7A DAI1 EQ5 */ - 0x00, /* 7B DAI1 EQ5 */ - 0x00, /* 7C DAI1 EQ5 */ - 0x00, /* 7D DAI1 EQ5 */ - 0x00, /* 7E DAI1 EQ5 */ - 0x00, /* 7F DAI1 EQ5 */ - - 0x00, /* 80 DAI1 EQ5 */ - 0x00, /* 81 DAI1 EQ5 */ - 0x00, /* 82 DAI1 EQ5 */ - 0x00, /* 83 DAI1 EQ5 */ - 0x00, /* 84 DAI2 EQ1 */ - 0x00, /* 85 DAI2 EQ1 */ - 0x00, /* 86 DAI2 EQ1 */ - 0x00, /* 87 DAI2 EQ1 */ - 0x00, /* 88 DAI2 EQ1 */ - 0x00, /* 89 DAI2 EQ1 */ - 0x00, /* 8A DAI2 EQ1 */ - 0x00, /* 8B DAI2 EQ1 */ - 0x00, /* 8C DAI2 EQ1 */ - 0x00, /* 8D DAI2 EQ1 */ - 0x00, /* 8E DAI2 EQ2 */ - 0x00, /* 8F DAI2 EQ2 */ - - 0x00, /* 90 DAI2 EQ2 */ - 0x00, /* 91 DAI2 EQ2 */ - 0x00, /* 92 DAI2 EQ2 */ - 0x00, /* 93 DAI2 EQ2 */ - 0x00, /* 94 DAI2 EQ2 */ - 0x00, /* 95 DAI2 EQ2 */ - 0x00, /* 96 DAI2 EQ2 */ - 0x00, /* 97 DAI2 EQ2 */ - 0x00, /* 98 DAI2 EQ3 */ - 0x00, /* 99 DAI2 EQ3 */ - 0x00, /* 9A DAI2 EQ3 */ - 0x00, /* 9B DAI2 EQ3 */ - 0x00, /* 9C DAI2 EQ3 */ - 0x00, /* 9D DAI2 EQ3 */ - 0x00, /* 9E DAI2 EQ3 */ - 0x00, /* 9F DAI2 EQ3 */ - - 0x00, /* A0 DAI2 EQ3 */ - 0x00, /* A1 DAI2 EQ3 */ - 0x00, /* A2 DAI2 EQ4 */ - 0x00, /* A3 DAI2 EQ4 */ - 0x00, /* A4 DAI2 EQ4 */ - 0x00, /* A5 DAI2 EQ4 */ - 0x00, /* A6 DAI2 EQ4 */ - 0x00, /* A7 DAI2 EQ4 */ - 0x00, /* A8 DAI2 EQ4 */ - 0x00, /* A9 DAI2 EQ4 */ - 0x00, /* AA DAI2 EQ4 */ - 0x00, /* AB DAI2 EQ4 */ - 0x00, /* AC DAI2 EQ5 */ - 0x00, /* AD DAI2 EQ5 */ - 0x00, /* AE DAI2 EQ5 */ - 0x00, /* AF DAI2 EQ5 */ - - 0x00, /* B0 DAI2 EQ5 */ - 0x00, /* B1 DAI2 EQ5 */ - 0x00, /* B2 DAI2 EQ5 */ - 0x00, /* B3 DAI2 EQ5 */ - 0x00, /* B4 DAI2 EQ5 */ - 0x00, /* B5 DAI2 EQ5 */ - 0x00, /* B6 DAI1 biquad */ - 0x00, /* B7 DAI1 biquad */ - 0x00, /* B8 DAI1 biquad */ - 0x00, /* B9 DAI1 biquad */ - 0x00, /* BA DAI1 biquad */ - 0x00, /* BB DAI1 biquad */ - 0x00, /* BC DAI1 biquad */ - 0x00, /* BD DAI1 biquad */ - 0x00, /* BE DAI1 biquad */ - 0x00, /* BF DAI1 biquad */ - - 0x00, /* C0 DAI2 biquad */ - 0x00, /* C1 DAI2 biquad */ - 0x00, /* C2 DAI2 biquad */ - 0x00, /* C3 DAI2 biquad */ - 0x00, /* C4 DAI2 biquad */ - 0x00, /* C5 DAI2 biquad */ - 0x00, /* C6 DAI2 biquad */ - 0x00, /* C7 DAI2 biquad */ - 0x00, /* C8 DAI2 biquad */ - 0x00, /* C9 DAI2 biquad */ - 0x00, /* CA */ - 0x00, /* CB */ - 0x00, /* CC */ - 0x00, /* CD */ - 0x00, /* CE */ - 0x00, /* CF */ - - 0x00, /* D0 */ - 0x00, /* D1 */ - 0x00, /* D2 */ - 0x00, /* D3 */ - 0x00, /* D4 */ - 0x00, /* D5 */ - 0x00, /* D6 */ - 0x00, /* D7 */ - 0x00, /* D8 */ - 0x00, /* D9 */ - 0x00, /* DA */ - 0x70, /* DB */ - 0x00, /* DC */ - 0x00, /* DD */ - 0x00, /* DE */ - 0x00, /* DF */ - - 0x00, /* E0 */ - 0x00, /* E1 */ - 0x00, /* E2 */ - 0x00, /* E3 */ - 0x00, /* E4 */ - 0x00, /* E5 */ - 0x00, /* E6 */ - 0x00, /* E7 */ - 0x00, /* E8 */ - 0x00, /* E9 */ - 0x00, /* EA */ - 0x00, /* EB */ - 0x00, /* EC */ - 0x00, /* ED */ - 0x00, /* EE */ - 0x00, /* EF */ - - 0x00, /* F0 */ - 0x00, /* F1 */ - 0x00, /* F2 */ - 0x00, /* F3 */ - 0x00, /* F4 */ - 0x00, /* F5 */ - 0x00, /* F6 */ - 0x00, /* F7 */ - 0x00, /* F8 */ - 0x00, /* F9 */ - 0x00, /* FA */ - 0x00, /* FB */ - 0x00, /* FC */ - 0x00, /* FD */ - 0x00, /* FE */ - 0x00, /* FF */ +static const struct reg_default max98088_reg[] = { + { 0xf, 0x00 }, /* 0F interrupt enable */ + + { 0x10, 0x00 }, /* 10 master clock */ + { 0x11, 0x00 }, /* 11 DAI1 clock mode */ + { 0x12, 0x00 }, /* 12 DAI1 clock control */ + { 0x13, 0x00 }, /* 13 DAI1 clock control */ + { 0x14, 0x00 }, /* 14 DAI1 format */ + { 0x15, 0x00 }, /* 15 DAI1 clock */ + { 0x16, 0x00 }, /* 16 DAI1 config */ + { 0x17, 0x00 }, /* 17 DAI1 TDM */ + { 0x18, 0x00 }, /* 18 DAI1 filters */ + { 0x19, 0x00 }, /* 19 DAI2 clock mode */ + { 0x1a, 0x00 }, /* 1A DAI2 clock control */ + { 0x1b, 0x00 }, /* 1B DAI2 clock control */ + { 0x1c, 0x00 }, /* 1C DAI2 format */ + { 0x1d, 0x00 }, /* 1D DAI2 clock */ + { 0x1e, 0x00 }, /* 1E DAI2 config */ + { 0x1f, 0x00 }, /* 1F DAI2 TDM */ + + { 0x20, 0x00 }, /* 20 DAI2 filters */ + { 0x21, 0x00 }, /* 21 data config */ + { 0x22, 0x00 }, /* 22 DAC mixer */ + { 0x23, 0x00 }, /* 23 left ADC mixer */ + { 0x24, 0x00 }, /* 24 right ADC mixer */ + { 0x25, 0x00 }, /* 25 left HP mixer */ + { 0x26, 0x00 }, /* 26 right HP mixer */ + { 0x27, 0x00 }, /* 27 HP control */ + { 0x28, 0x00 }, /* 28 left REC mixer */ + { 0x29, 0x00 }, /* 29 right REC mixer */ + { 0x2a, 0x00 }, /* 2A REC control */ + { 0x2b, 0x00 }, /* 2B left SPK mixer */ + { 0x2c, 0x00 }, /* 2C right SPK mixer */ + { 0x2d, 0x00 }, /* 2D SPK control */ + { 0x2e, 0x00 }, /* 2E sidetone */ + { 0x2f, 0x00 }, /* 2F DAI1 playback level */ + + { 0x30, 0x00 }, /* 30 DAI1 playback level */ + { 0x31, 0x00 }, /* 31 DAI2 playback level */ + { 0x32, 0x00 }, /* 32 DAI2 playbakc level */ + { 0x33, 0x00 }, /* 33 left ADC level */ + { 0x34, 0x00 }, /* 34 right ADC level */ + { 0x35, 0x00 }, /* 35 MIC1 level */ + { 0x36, 0x00 }, /* 36 MIC2 level */ + { 0x37, 0x00 }, /* 37 INA level */ + { 0x38, 0x00 }, /* 38 INB level */ + { 0x39, 0x00 }, /* 39 left HP volume */ + { 0x3a, 0x00 }, /* 3A right HP volume */ + { 0x3b, 0x00 }, /* 3B left REC volume */ + { 0x3c, 0x00 }, /* 3C right REC volume */ + { 0x3d, 0x00 }, /* 3D left SPK volume */ + { 0x3e, 0x00 }, /* 3E right SPK volume */ + { 0x3f, 0x00 }, /* 3F MIC config */ + + { 0x40, 0x00 }, /* 40 MIC threshold */ + { 0x41, 0x00 }, /* 41 excursion limiter filter */ + { 0x42, 0x00 }, /* 42 excursion limiter threshold */ + { 0x43, 0x00 }, /* 43 ALC */ + { 0x44, 0x00 }, /* 44 power limiter threshold */ + { 0x45, 0x00 }, /* 45 power limiter config */ + { 0x46, 0x00 }, /* 46 distortion limiter config */ + { 0x47, 0x00 }, /* 47 audio input */ + { 0x48, 0x00 }, /* 48 microphone */ + { 0x49, 0x00 }, /* 49 level control */ + { 0x4a, 0x00 }, /* 4A bypass switches */ + { 0x4b, 0x00 }, /* 4B jack detect */ + { 0x4c, 0x00 }, /* 4C input enable */ + { 0x4d, 0x00 }, /* 4D output enable */ + { 0x4e, 0xF0 }, /* 4E bias control */ + { 0x4f, 0x00 }, /* 4F DAC power */ + + { 0x50, 0x0F }, /* 50 DAC power */ + { 0x51, 0x00 }, /* 51 system */ + { 0x52, 0x00 }, /* 52 DAI1 EQ1 */ + { 0x53, 0x00 }, /* 53 DAI1 EQ1 */ + { 0x54, 0x00 }, /* 54 DAI1 EQ1 */ + { 0x55, 0x00 }, /* 55 DAI1 EQ1 */ + { 0x56, 0x00 }, /* 56 DAI1 EQ1 */ + { 0x57, 0x00 }, /* 57 DAI1 EQ1 */ + { 0x58, 0x00 }, /* 58 DAI1 EQ1 */ + { 0x59, 0x00 }, /* 59 DAI1 EQ1 */ + { 0x5a, 0x00 }, /* 5A DAI1 EQ1 */ + { 0x5b, 0x00 }, /* 5B DAI1 EQ1 */ + { 0x5c, 0x00 }, /* 5C DAI1 EQ2 */ + { 0x5d, 0x00 }, /* 5D DAI1 EQ2 */ + { 0x5e, 0x00 }, /* 5E DAI1 EQ2 */ + { 0x5f, 0x00 }, /* 5F DAI1 EQ2 */ + + { 0x60, 0x00 }, /* 60 DAI1 EQ2 */ + { 0x61, 0x00 }, /* 61 DAI1 EQ2 */ + { 0x62, 0x00 }, /* 62 DAI1 EQ2 */ + { 0x63, 0x00 }, /* 63 DAI1 EQ2 */ + { 0x64, 0x00 }, /* 64 DAI1 EQ2 */ + { 0x65, 0x00 }, /* 65 DAI1 EQ2 */ + { 0x66, 0x00 }, /* 66 DAI1 EQ3 */ + { 0x67, 0x00 }, /* 67 DAI1 EQ3 */ + { 0x68, 0x00 }, /* 68 DAI1 EQ3 */ + { 0x69, 0x00 }, /* 69 DAI1 EQ3 */ + { 0x6a, 0x00 }, /* 6A DAI1 EQ3 */ + { 0x6b, 0x00 }, /* 6B DAI1 EQ3 */ + { 0x6c, 0x00 }, /* 6C DAI1 EQ3 */ + { 0x6d, 0x00 }, /* 6D DAI1 EQ3 */ + { 0x6e, 0x00 }, /* 6E DAI1 EQ3 */ + { 0x6f, 0x00 }, /* 6F DAI1 EQ3 */ + + { 0x70, 0x00 }, /* 70 DAI1 EQ4 */ + { 0x71, 0x00 }, /* 71 DAI1 EQ4 */ + { 0x72, 0x00 }, /* 72 DAI1 EQ4 */ + { 0x73, 0x00 }, /* 73 DAI1 EQ4 */ + { 0x74, 0x00 }, /* 74 DAI1 EQ4 */ + { 0x75, 0x00 }, /* 75 DAI1 EQ4 */ + { 0x76, 0x00 }, /* 76 DAI1 EQ4 */ + { 0x77, 0x00 }, /* 77 DAI1 EQ4 */ + { 0x78, 0x00 }, /* 78 DAI1 EQ4 */ + { 0x79, 0x00 }, /* 79 DAI1 EQ4 */ + { 0x7a, 0x00 }, /* 7A DAI1 EQ5 */ + { 0x7b, 0x00 }, /* 7B DAI1 EQ5 */ + { 0x7c, 0x00 }, /* 7C DAI1 EQ5 */ + { 0x7d, 0x00 }, /* 7D DAI1 EQ5 */ + { 0x7e, 0x00 }, /* 7E DAI1 EQ5 */ + { 0x7f, 0x00 }, /* 7F DAI1 EQ5 */ + + { 0x80, 0x00 }, /* 80 DAI1 EQ5 */ + { 0x81, 0x00 }, /* 81 DAI1 EQ5 */ + { 0x82, 0x00 }, /* 82 DAI1 EQ5 */ + { 0x83, 0x00 }, /* 83 DAI1 EQ5 */ + { 0x84, 0x00 }, /* 84 DAI2 EQ1 */ + { 0x85, 0x00 }, /* 85 DAI2 EQ1 */ + { 0x86, 0x00 }, /* 86 DAI2 EQ1 */ + { 0x87, 0x00 }, /* 87 DAI2 EQ1 */ + { 0x88, 0x00 }, /* 88 DAI2 EQ1 */ + { 0x89, 0x00 }, /* 89 DAI2 EQ1 */ + { 0x8a, 0x00 }, /* 8A DAI2 EQ1 */ + { 0x8b, 0x00 }, /* 8B DAI2 EQ1 */ + { 0x8c, 0x00 }, /* 8C DAI2 EQ1 */ + { 0x8d, 0x00 }, /* 8D DAI2 EQ1 */ + { 0x8e, 0x00 }, /* 8E DAI2 EQ2 */ + { 0x8f, 0x00 }, /* 8F DAI2 EQ2 */ + + { 0x90, 0x00 }, /* 90 DAI2 EQ2 */ + { 0x91, 0x00 }, /* 91 DAI2 EQ2 */ + { 0x92, 0x00 }, /* 92 DAI2 EQ2 */ + { 0x93, 0x00 }, /* 93 DAI2 EQ2 */ + { 0x94, 0x00 }, /* 94 DAI2 EQ2 */ + { 0x95, 0x00 }, /* 95 DAI2 EQ2 */ + { 0x96, 0x00 }, /* 96 DAI2 EQ2 */ + { 0x97, 0x00 }, /* 97 DAI2 EQ2 */ + { 0x98, 0x00 }, /* 98 DAI2 EQ3 */ + { 0x99, 0x00 }, /* 99 DAI2 EQ3 */ + { 0x9a, 0x00 }, /* 9A DAI2 EQ3 */ + { 0x9b, 0x00 }, /* 9B DAI2 EQ3 */ + { 0x9c, 0x00 }, /* 9C DAI2 EQ3 */ + { 0x9d, 0x00 }, /* 9D DAI2 EQ3 */ + { 0x9e, 0x00 }, /* 9E DAI2 EQ3 */ + { 0x9f, 0x00 }, /* 9F DAI2 EQ3 */ + + { 0xa0, 0x00 }, /* A0 DAI2 EQ3 */ + { 0xa1, 0x00 }, /* A1 DAI2 EQ3 */ + { 0xa2, 0x00 }, /* A2 DAI2 EQ4 */ + { 0xa3, 0x00 }, /* A3 DAI2 EQ4 */ + { 0xa4, 0x00 }, /* A4 DAI2 EQ4 */ + { 0xa5, 0x00 }, /* A5 DAI2 EQ4 */ + { 0xa6, 0x00 }, /* A6 DAI2 EQ4 */ + { 0xa7, 0x00 }, /* A7 DAI2 EQ4 */ + { 0xa8, 0x00 }, /* A8 DAI2 EQ4 */ + { 0xa9, 0x00 }, /* A9 DAI2 EQ4 */ + { 0xaa, 0x00 }, /* AA DAI2 EQ4 */ + { 0xab, 0x00 }, /* AB DAI2 EQ4 */ + { 0xac, 0x00 }, /* AC DAI2 EQ5 */ + { 0xad, 0x00 }, /* AD DAI2 EQ5 */ + { 0xae, 0x00 }, /* AE DAI2 EQ5 */ + { 0xaf, 0x00 }, /* AF DAI2 EQ5 */ + + { 0xb0, 0x00 }, /* B0 DAI2 EQ5 */ + { 0xb1, 0x00 }, /* B1 DAI2 EQ5 */ + { 0xb2, 0x00 }, /* B2 DAI2 EQ5 */ + { 0xb3, 0x00 }, /* B3 DAI2 EQ5 */ + { 0xb4, 0x00 }, /* B4 DAI2 EQ5 */ + { 0xb5, 0x00 }, /* B5 DAI2 EQ5 */ + { 0xb6, 0x00 }, /* B6 DAI1 biquad */ + { 0xb7, 0x00 }, /* B7 DAI1 biquad */ + { 0xb8 ,0x00 }, /* B8 DAI1 biquad */ + { 0xb9, 0x00 }, /* B9 DAI1 biquad */ + { 0xba, 0x00 }, /* BA DAI1 biquad */ + { 0xbb, 0x00 }, /* BB DAI1 biquad */ + { 0xbc, 0x00 }, /* BC DAI1 biquad */ + { 0xbd, 0x00 }, /* BD DAI1 biquad */ + { 0xbe, 0x00 }, /* BE DAI1 biquad */ + { 0xbf, 0x00 }, /* BF DAI1 biquad */ + + { 0xc0, 0x00 }, /* C0 DAI2 biquad */ + { 0xc1, 0x00 }, /* C1 DAI2 biquad */ + { 0xc2, 0x00 }, /* C2 DAI2 biquad */ + { 0xc3, 0x00 }, /* C3 DAI2 biquad */ + { 0xc4, 0x00 }, /* C4 DAI2 biquad */ + { 0xc5, 0x00 }, /* C5 DAI2 biquad */ + { 0xc6, 0x00 }, /* C6 DAI2 biquad */ + { 0xc7, 0x00 }, /* C7 DAI2 biquad */ + { 0xc8, 0x00 }, /* C8 DAI2 biquad */ + { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; static struct { @@ -606,11 +536,27 @@ static struct { { 0xFF, 0x00, 1 }, /* FF */ }; -static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool max98088_readable_register(struct device *dev, unsigned int reg) +{ + return max98088_access[reg].readable; +} + +static bool max98088_volatile_register(struct device *dev, unsigned int reg) { return max98088_access[reg].vol; } +static const struct regmap_config max98088_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max98088_readable_register, + .volatile_reg = max98088_volatile_register, + + .reg_defaults = max98088_reg, + .num_reg_defaults = ARRAY_SIZE(max98088_reg), + .cache_type = REGCACHE_RBTREE, +}; /* * Load equalizer DSP coefficient configurations registers @@ -1610,58 +1556,34 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static void max98088_sync_cache(struct snd_soc_codec *codec) -{ - u8 *reg_cache = codec->reg_cache; - int i; - - if (!codec->cache_sync) - return; - - codec->cache_only = 0; - - /* write back cached values if they're writeable and - * different from the hardware default. - */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - if (!max98088_access[i].writable) - continue; - - if (reg_cache[i] == max98088_reg[i]) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } - - codec->cache_sync = 0; -} - static int max98088_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - break; - - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - max98088_sync_cache(codec); - - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, - M98088_MBEN, M98088_MBEN); - break; - - case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, - M98088_MBEN, 0); - codec->cache_sync = 1; - break; - } - codec->dapm.bias_level = level; - return 0; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(max98088->regmap); + + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, M98088_MBEN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, 0); + regcache_mark_dirty(max98088->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; } #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 @@ -1988,9 +1910,9 @@ static int max98088_probe(struct snd_soc_codec *codec) struct max98088_cdata *cdata; int ret = 0; - codec->cache_sync = 1; + regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + 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; @@ -2070,10 +1992,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { .set_bias_level = max98088_set_bias_level, .controls = max98088_snd_controls, .num_controls = ARRAY_SIZE(max98088_snd_controls), - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, @@ -2081,7 +1999,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { }; static int max98088_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { struct max98088_priv *max98088; int ret; @@ -2091,6 +2009,10 @@ static int max98088_i2c_probe(struct i2c_client *i2c, if (max98088 == NULL) return -ENOMEM; + max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); + if (IS_ERR(max98088->regmap)) + return PTR_ERR(max98088->regmap); + max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); -- cgit From 2245e3c31c15c2d2a26926c4b734f4d3a37ae252 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 11:50:10 +0100 Subject: ASoC: ab8500: Explicitly set I/O up We do some I/O in probe so we need to ensure the I/O operations are fully set up then. Reported-by: Olof Johansson Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index d5a0fc4b2fe2..7f6ca111659b 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2468,6 +2468,8 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- cgit From d36126ac5674a83e1d426877709437b24f058f47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 18:58:59 +0100 Subject: ASoC: max98095: Remove custom hw_write() implementation The registers that are being kept uncached are marked as volatile anyway so the call has no practical impact. Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd1642970..65aba5ec52df 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -611,23 +611,6 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) return 0; } -/* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - - codec->cache_bypass = 1; - ret = snd_soc_write(codec, reg, value); - codec->cache_bypass = 0; - - return ret ? -EIO : 0; -} - /* * Load equalizer DSP coefficient configurations registers */ @@ -648,8 +631,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -673,8 +656,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } -- cgit From c6b3283f6d8818177349eb7cc0549286a55140c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:05:16 +0100 Subject: ASoC: max90895: Convert to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 65aba5ec52df..1a4585ae36e4 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1268,14 +1268,6 @@ static const struct snd_soc_dapm_route max98095_audio_map[] = { {"MIC2 Input", NULL, "MIC2"}, }; -static int max98095_add_widgets(struct snd_soc_codec *codec) -{ - snd_soc_add_codec_controls(codec, max98095_snd_controls, - ARRAY_SIZE(max98095_snd_controls)); - - return 0; -} - /* codec mclk clock divider coefficients */ static const struct { u32 rate; @@ -2430,8 +2422,6 @@ static int max98095_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98095_097_PWR_SYS, M98095_SHDNRUN, M98095_SHDNRUN); - max98095_add_widgets(codec); - return 0; err_irq: @@ -2463,6 +2453,8 @@ static struct snd_soc_codec_driver soc_codec_dev_max98095 = { .suspend = max98095_suspend, .resume = max98095_resume, .set_bias_level = max98095_set_bias_level, + .controls = max98095_snd_controls, + .num_controls = ARRAY_SIZE(max98095_snd_controls), .reg_cache_size = ARRAY_SIZE(max98095_reg_def), .reg_word_size = sizeof(u8), .reg_cache_default = max98095_reg_def, -- cgit From 14acbbbbc649c4c6057f601396b8000cd616d9ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:08:35 +0100 Subject: ASoC: max98095: Convert to direct regmap API usage Saves code and moves us towards being able to remove the duplicate ASoC level register I/O functionality. Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 435 +++++++++++++++++--------------------------- 1 file changed, 167 insertions(+), 268 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 1a4585ae36e4..5c9f6b527cf0 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -39,6 +39,7 @@ struct max98095_cdata { }; struct max98095_priv { + struct regmap *regmap; enum max98095_type devtype; struct max98095_pdata *pdata; unsigned int sysclk; @@ -56,263 +57,145 @@ struct max98095_priv { struct snd_soc_jack *mic_jack; }; -static const u8 max98095_reg_def[M98095_REG_CNT] = { - 0x00, /* 00 */ - 0x00, /* 01 */ - 0x00, /* 02 */ - 0x00, /* 03 */ - 0x00, /* 04 */ - 0x00, /* 05 */ - 0x00, /* 06 */ - 0x00, /* 07 */ - 0x00, /* 08 */ - 0x00, /* 09 */ - 0x00, /* 0A */ - 0x00, /* 0B */ - 0x00, /* 0C */ - 0x00, /* 0D */ - 0x00, /* 0E */ - 0x00, /* 0F */ - 0x00, /* 10 */ - 0x00, /* 11 */ - 0x00, /* 12 */ - 0x00, /* 13 */ - 0x00, /* 14 */ - 0x00, /* 15 */ - 0x00, /* 16 */ - 0x00, /* 17 */ - 0x00, /* 18 */ - 0x00, /* 19 */ - 0x00, /* 1A */ - 0x00, /* 1B */ - 0x00, /* 1C */ - 0x00, /* 1D */ - 0x00, /* 1E */ - 0x00, /* 1F */ - 0x00, /* 20 */ - 0x00, /* 21 */ - 0x00, /* 22 */ - 0x00, /* 23 */ - 0x00, /* 24 */ - 0x00, /* 25 */ - 0x00, /* 26 */ - 0x00, /* 27 */ - 0x00, /* 28 */ - 0x00, /* 29 */ - 0x00, /* 2A */ - 0x00, /* 2B */ - 0x00, /* 2C */ - 0x00, /* 2D */ - 0x00, /* 2E */ - 0x00, /* 2F */ - 0x00, /* 30 */ - 0x00, /* 31 */ - 0x00, /* 32 */ - 0x00, /* 33 */ - 0x00, /* 34 */ - 0x00, /* 35 */ - 0x00, /* 36 */ - 0x00, /* 37 */ - 0x00, /* 38 */ - 0x00, /* 39 */ - 0x00, /* 3A */ - 0x00, /* 3B */ - 0x00, /* 3C */ - 0x00, /* 3D */ - 0x00, /* 3E */ - 0x00, /* 3F */ - 0x00, /* 40 */ - 0x00, /* 41 */ - 0x00, /* 42 */ - 0x00, /* 43 */ - 0x00, /* 44 */ - 0x00, /* 45 */ - 0x00, /* 46 */ - 0x00, /* 47 */ - 0x00, /* 48 */ - 0x00, /* 49 */ - 0x00, /* 4A */ - 0x00, /* 4B */ - 0x00, /* 4C */ - 0x00, /* 4D */ - 0x00, /* 4E */ - 0x00, /* 4F */ - 0x00, /* 50 */ - 0x00, /* 51 */ - 0x00, /* 52 */ - 0x00, /* 53 */ - 0x00, /* 54 */ - 0x00, /* 55 */ - 0x00, /* 56 */ - 0x00, /* 57 */ - 0x00, /* 58 */ - 0x00, /* 59 */ - 0x00, /* 5A */ - 0x00, /* 5B */ - 0x00, /* 5C */ - 0x00, /* 5D */ - 0x00, /* 5E */ - 0x00, /* 5F */ - 0x00, /* 60 */ - 0x00, /* 61 */ - 0x00, /* 62 */ - 0x00, /* 63 */ - 0x00, /* 64 */ - 0x00, /* 65 */ - 0x00, /* 66 */ - 0x00, /* 67 */ - 0x00, /* 68 */ - 0x00, /* 69 */ - 0x00, /* 6A */ - 0x00, /* 6B */ - 0x00, /* 6C */ - 0x00, /* 6D */ - 0x00, /* 6E */ - 0x00, /* 6F */ - 0x00, /* 70 */ - 0x00, /* 71 */ - 0x00, /* 72 */ - 0x00, /* 73 */ - 0x00, /* 74 */ - 0x00, /* 75 */ - 0x00, /* 76 */ - 0x00, /* 77 */ - 0x00, /* 78 */ - 0x00, /* 79 */ - 0x00, /* 7A */ - 0x00, /* 7B */ - 0x00, /* 7C */ - 0x00, /* 7D */ - 0x00, /* 7E */ - 0x00, /* 7F */ - 0x00, /* 80 */ - 0x00, /* 81 */ - 0x00, /* 82 */ - 0x00, /* 83 */ - 0x00, /* 84 */ - 0x00, /* 85 */ - 0x00, /* 86 */ - 0x00, /* 87 */ - 0x00, /* 88 */ - 0x00, /* 89 */ - 0x00, /* 8A */ - 0x00, /* 8B */ - 0x00, /* 8C */ - 0x00, /* 8D */ - 0x00, /* 8E */ - 0x00, /* 8F */ - 0x00, /* 90 */ - 0x00, /* 91 */ - 0x30, /* 92 */ - 0xF0, /* 93 */ - 0x00, /* 94 */ - 0x00, /* 95 */ - 0x3F, /* 96 */ - 0x00, /* 97 */ - 0x00, /* 98 */ - 0x00, /* 99 */ - 0x00, /* 9A */ - 0x00, /* 9B */ - 0x00, /* 9C */ - 0x00, /* 9D */ - 0x00, /* 9E */ - 0x00, /* 9F */ - 0x00, /* A0 */ - 0x00, /* A1 */ - 0x00, /* A2 */ - 0x00, /* A3 */ - 0x00, /* A4 */ - 0x00, /* A5 */ - 0x00, /* A6 */ - 0x00, /* A7 */ - 0x00, /* A8 */ - 0x00, /* A9 */ - 0x00, /* AA */ - 0x00, /* AB */ - 0x00, /* AC */ - 0x00, /* AD */ - 0x00, /* AE */ - 0x00, /* AF */ - 0x00, /* B0 */ - 0x00, /* B1 */ - 0x00, /* B2 */ - 0x00, /* B3 */ - 0x00, /* B4 */ - 0x00, /* B5 */ - 0x00, /* B6 */ - 0x00, /* B7 */ - 0x00, /* B8 */ - 0x00, /* B9 */ - 0x00, /* BA */ - 0x00, /* BB */ - 0x00, /* BC */ - 0x00, /* BD */ - 0x00, /* BE */ - 0x00, /* BF */ - 0x00, /* C0 */ - 0x00, /* C1 */ - 0x00, /* C2 */ - 0x00, /* C3 */ - 0x00, /* C4 */ - 0x00, /* C5 */ - 0x00, /* C6 */ - 0x00, /* C7 */ - 0x00, /* C8 */ - 0x00, /* C9 */ - 0x00, /* CA */ - 0x00, /* CB */ - 0x00, /* CC */ - 0x00, /* CD */ - 0x00, /* CE */ - 0x00, /* CF */ - 0x00, /* D0 */ - 0x00, /* D1 */ - 0x00, /* D2 */ - 0x00, /* D3 */ - 0x00, /* D4 */ - 0x00, /* D5 */ - 0x00, /* D6 */ - 0x00, /* D7 */ - 0x00, /* D8 */ - 0x00, /* D9 */ - 0x00, /* DA */ - 0x00, /* DB */ - 0x00, /* DC */ - 0x00, /* DD */ - 0x00, /* DE */ - 0x00, /* DF */ - 0x00, /* E0 */ - 0x00, /* E1 */ - 0x00, /* E2 */ - 0x00, /* E3 */ - 0x00, /* E4 */ - 0x00, /* E5 */ - 0x00, /* E6 */ - 0x00, /* E7 */ - 0x00, /* E8 */ - 0x00, /* E9 */ - 0x00, /* EA */ - 0x00, /* EB */ - 0x00, /* EC */ - 0x00, /* ED */ - 0x00, /* EE */ - 0x00, /* EF */ - 0x00, /* F0 */ - 0x00, /* F1 */ - 0x00, /* F2 */ - 0x00, /* F3 */ - 0x00, /* F4 */ - 0x00, /* F5 */ - 0x00, /* F6 */ - 0x00, /* F7 */ - 0x00, /* F8 */ - 0x00, /* F9 */ - 0x00, /* FA */ - 0x00, /* FB */ - 0x00, /* FC */ - 0x00, /* FD */ - 0x00, /* FE */ - 0x00, /* FF */ +static const struct reg_default max98095_reg_def[] = { + { 0xf, 0x00 }, /* 0F */ + { 0x10, 0x00 }, /* 10 */ + { 0x11, 0x00 }, /* 11 */ + { 0x12, 0x00 }, /* 12 */ + { 0x13, 0x00 }, /* 13 */ + { 0x14, 0x00 }, /* 14 */ + { 0x15, 0x00 }, /* 15 */ + { 0x16, 0x00 }, /* 16 */ + { 0x17, 0x00 }, /* 17 */ + { 0x18, 0x00 }, /* 18 */ + { 0x19, 0x00 }, /* 19 */ + { 0x1a, 0x00 }, /* 1A */ + { 0x1b, 0x00 }, /* 1B */ + { 0x1c, 0x00 }, /* 1C */ + { 0x1d, 0x00 }, /* 1D */ + { 0x1e, 0x00 }, /* 1E */ + { 0x1f, 0x00 }, /* 1F */ + { 0x20, 0x00 }, /* 20 */ + { 0x21, 0x00 }, /* 21 */ + { 0x22, 0x00 }, /* 22 */ + { 0x23, 0x00 }, /* 23 */ + { 0x24, 0x00 }, /* 24 */ + { 0x25, 0x00 }, /* 25 */ + { 0x26, 0x00 }, /* 26 */ + { 0x27, 0x00 }, /* 27 */ + { 0x28, 0x00 }, /* 28 */ + { 0x29, 0x00 }, /* 29 */ + { 0x2a, 0x00 }, /* 2A */ + { 0x2b, 0x00 }, /* 2B */ + { 0x2c, 0x00 }, /* 2C */ + { 0x2d, 0x00 }, /* 2D */ + { 0x2e, 0x00 }, /* 2E */ + { 0x2f, 0x00 }, /* 2F */ + { 0x30, 0x00 }, /* 30 */ + { 0x31, 0x00 }, /* 31 */ + { 0x32, 0x00 }, /* 32 */ + { 0x33, 0x00 }, /* 33 */ + { 0x34, 0x00 }, /* 34 */ + { 0x35, 0x00 }, /* 35 */ + { 0x36, 0x00 }, /* 36 */ + { 0x37, 0x00 }, /* 37 */ + { 0x38, 0x00 }, /* 38 */ + { 0x39, 0x00 }, /* 39 */ + { 0x3a, 0x00 }, /* 3A */ + { 0x3b, 0x00 }, /* 3B */ + { 0x3c, 0x00 }, /* 3C */ + { 0x3d, 0x00 }, /* 3D */ + { 0x3e, 0x00 }, /* 3E */ + { 0x3f, 0x00 }, /* 3F */ + { 0x40, 0x00 }, /* 40 */ + { 0x41, 0x00 }, /* 41 */ + { 0x42, 0x00 }, /* 42 */ + { 0x43, 0x00 }, /* 43 */ + { 0x44, 0x00 }, /* 44 */ + { 0x45, 0x00 }, /* 45 */ + { 0x46, 0x00 }, /* 46 */ + { 0x47, 0x00 }, /* 47 */ + { 0x48, 0x00 }, /* 48 */ + { 0x49, 0x00 }, /* 49 */ + { 0x4a, 0x00 }, /* 4A */ + { 0x4b, 0x00 }, /* 4B */ + { 0x4c, 0x00 }, /* 4C */ + { 0x4d, 0x00 }, /* 4D */ + { 0x4e, 0x00 }, /* 4E */ + { 0x4f, 0x00 }, /* 4F */ + { 0x50, 0x00 }, /* 50 */ + { 0x51, 0x00 }, /* 51 */ + { 0x52, 0x00 }, /* 52 */ + { 0x53, 0x00 }, /* 53 */ + { 0x54, 0x00 }, /* 54 */ + { 0x55, 0x00 }, /* 55 */ + { 0x56, 0x00 }, /* 56 */ + { 0x57, 0x00 }, /* 57 */ + { 0x58, 0x00 }, /* 58 */ + { 0x59, 0x00 }, /* 59 */ + { 0x5a, 0x00 }, /* 5A */ + { 0x5b, 0x00 }, /* 5B */ + { 0x5c, 0x00 }, /* 5C */ + { 0x5d, 0x00 }, /* 5D */ + { 0x5e, 0x00 }, /* 5E */ + { 0x5f, 0x00 }, /* 5F */ + { 0x60, 0x00 }, /* 60 */ + { 0x61, 0x00 }, /* 61 */ + { 0x62, 0x00 }, /* 62 */ + { 0x63, 0x00 }, /* 63 */ + { 0x64, 0x00 }, /* 64 */ + { 0x65, 0x00 }, /* 65 */ + { 0x66, 0x00 }, /* 66 */ + { 0x67, 0x00 }, /* 67 */ + { 0x68, 0x00 }, /* 68 */ + { 0x69, 0x00 }, /* 69 */ + { 0x6a, 0x00 }, /* 6A */ + { 0x6b, 0x00 }, /* 6B */ + { 0x6c, 0x00 }, /* 6C */ + { 0x6d, 0x00 }, /* 6D */ + { 0x6e, 0x00 }, /* 6E */ + { 0x6f, 0x00 }, /* 6F */ + { 0x70, 0x00 }, /* 70 */ + { 0x71, 0x00 }, /* 71 */ + { 0x72, 0x00 }, /* 72 */ + { 0x73, 0x00 }, /* 73 */ + { 0x74, 0x00 }, /* 74 */ + { 0x75, 0x00 }, /* 75 */ + { 0x76, 0x00 }, /* 76 */ + { 0x77, 0x00 }, /* 77 */ + { 0x78, 0x00 }, /* 78 */ + { 0x79, 0x00 }, /* 79 */ + { 0x7a, 0x00 }, /* 7A */ + { 0x7b, 0x00 }, /* 7B */ + { 0x7c, 0x00 }, /* 7C */ + { 0x7d, 0x00 }, /* 7D */ + { 0x7e, 0x00 }, /* 7E */ + { 0x7f, 0x00 }, /* 7F */ + { 0x80, 0x00 }, /* 80 */ + { 0x81, 0x00 }, /* 81 */ + { 0x82, 0x00 }, /* 82 */ + { 0x83, 0x00 }, /* 83 */ + { 0x84, 0x00 }, /* 84 */ + { 0x85, 0x00 }, /* 85 */ + { 0x86, 0x00 }, /* 86 */ + { 0x87, 0x00 }, /* 87 */ + { 0x88, 0x00 }, /* 88 */ + { 0x89, 0x00 }, /* 89 */ + { 0x8a, 0x00 }, /* 8A */ + { 0x8b, 0x00 }, /* 8B */ + { 0x8c, 0x00 }, /* 8C */ + { 0x8d, 0x00 }, /* 8D */ + { 0x8e, 0x00 }, /* 8E */ + { 0x8f, 0x00 }, /* 8F */ + { 0x90, 0x00 }, /* 90 */ + { 0x91, 0x00 }, /* 91 */ + { 0x92, 0x30 }, /* 92 */ + { 0x93, 0xF0 }, /* 93 */ + { 0x94, 0x00 }, /* 94 */ + { 0x95, 0x00 }, /* 95 */ + { 0x96, 0x3F }, /* 96 */ + { 0x97, 0x00 }, /* 97 */ + { 0xff, 0x00 }, /* FF */ }; static struct { @@ -577,14 +460,14 @@ static struct { { 0xFF, 0x00 }, /* FF */ }; -static int max98095_readable(struct snd_soc_codec *codec, unsigned int reg) +static bool max98095_readable(struct device *dev, unsigned int reg) { if (reg >= M98095_REG_CNT) return 0; return max98095_access[reg].readable != 0; } -static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool max98095_volatile(struct device *dev, unsigned int reg) { if (reg > M98095_REG_MAX_CACHED) return 1; @@ -611,6 +494,19 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) return 0; } +static const struct regmap_config max98095_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max98095_reg_def, + .num_reg_defaults = ARRAY_SIZE(max98095_reg_def), + .max_register = M98095_0FF_REV_ID, + .cache_type = REGCACHE_RBTREE, + + .readable_reg = max98095_readable, + .volatile_reg = max98095_volatile, +}; + /* * Load equalizer DSP coefficient configurations registers */ @@ -1723,6 +1619,7 @@ static int max98095_dai3_set_fmt(struct snd_soc_dai *codec_dai, static int max98095_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -1734,7 +1631,7 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max98095->regmap); if (ret != 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -1749,7 +1646,7 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, M98095_090_PWR_EN_IN, M98095_MBEN, 0); - codec->cache_sync = 1; + regcache_mark_dirty(max98095->regmap); break; } codec->dapm.bias_level = level; @@ -2316,7 +2213,7 @@ static int max98095_reset(struct snd_soc_codec *codec) /* Reset to hardware default for registers, as there is not * a soft reset hardware control register */ for (i = M98095_010_HOST_INT_CFG; i < M98095_REG_MAX_CACHED; i++) { - ret = snd_soc_write(codec, i, max98095_reg_def[i]); + ret = snd_soc_write(codec, i, snd_soc_read(codec, i)); if (ret < 0) { dev_err(codec->dev, "Failed to reset: %d\n", ret); return ret; @@ -2333,7 +2230,7 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + 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; @@ -2455,11 +2352,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max98095 = { .set_bias_level = max98095_set_bias_level, .controls = max98095_snd_controls, .num_controls = ARRAY_SIZE(max98095_snd_controls), - .reg_cache_size = ARRAY_SIZE(max98095_reg_def), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98095_reg_def, - .readable_register = max98095_readable, - .volatile_register = max98095_volatile, .dapm_widgets = max98095_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98095_dapm_widgets), .dapm_routes = max98095_audio_map, @@ -2477,6 +2369,13 @@ static int max98095_i2c_probe(struct i2c_client *i2c, if (max98095 == NULL) return -ENOMEM; + max98095->regmap = devm_regmap_init_i2c(i2c, &max98095_regmap); + if (IS_ERR(max98095->regmap)) { + ret = PTR_ERR(max98095->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + max98095->devtype = id->driver_data; i2c_set_clientdata(i2c, max98095); max98095->pdata = i2c->dev.platform_data; -- cgit From 2c142c61f79c14a120c0f4d2954e35b6404b2d0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 18:49:54 +0100 Subject: ASoC: tlv320aic23: Remove #defines for I2C The only control interface supported by this driver is I2C so there is no need for conditional compilation around the control interface. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 31762ebdd774..32994597a43f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -613,7 +613,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -660,29 +659,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .id_table = tlv320aic23_id, }; -#endif - -static int __init tlv320aic23_modinit(void) -{ - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&tlv320aic23_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(tlv320aic23_modinit); - -static void __exit tlv320aic23_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&tlv320aic23_i2c_driver); -#endif -} -module_exit(tlv320aic23_exit); +module_i2c_driver(tlv320aic23_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS "); -- cgit From b07c443fabb97f909c8cc406bfd2d0ecc002bc3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 18:51:26 +0100 Subject: ASoC: tlv320aic23: Convert to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 32994597a43f..3a6be8c3d557 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -586,9 +586,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); - snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls, - ARRAY_SIZE(tlv320aic23_snd_controls)); - return 0; } @@ -607,6 +604,8 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, .set_bias_level = tlv320aic23_set_bias_level, + .controls = tlv320aic23_snd_controls, + .num_controls = ARRAY_SIZE(tlv320aic23_snd_controls), .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), .dapm_routes = tlv320aic23_intercon, -- cgit From a16bbe4d685c1465b98d3fabdb95310eafcd383e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 20:07:12 +0100 Subject: ASoC: tlv320aic3x: Remove nonsense comment for register cache Every statement in this comment is incorrect either through bitrot or (mostly) through never having corresponded to reality in the first place. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e3f269243e0..3abbff3fe888 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -90,12 +90,6 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -/* - * AIC3X register cache - * We can't read the AIC3X register space when we are - * using 2 wire for device control, so we cache them instead. - * There is no point in caching the reset register - */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x10, /* 0 */ 0x04, 0x00, 0x00, 0x00, /* 4 */ -- cgit From 6f818e04fc8d3d413eeb3a689c7607f2a89ab568 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:48:45 +0100 Subject: ASoC: tlv320aic3x: Move resource acquisition to I2C probe This is more idiomatic and interacts better with deferred probing. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 63 ++++++++++++++++++++++-------------------- 1 file changed, 33 insertions(+), 30 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3abbff3fe888..de17a36beb6f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1345,23 +1345,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) return ret; } - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) { - ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); - if (ret != 0) - goto err_gpio; - gpio_direction_output(aic3x->gpio_reset, 0); - } - - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) - aic3x->supplies[i].supply = aic3x_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies), - aic3x->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err_get; - } for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; @@ -1418,12 +1401,6 @@ err_notif: while (i--) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); - regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); -err_get: - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) - gpio_free(aic3x->gpio_reset); -err_gpio: return ret; } @@ -1434,15 +1411,9 @@ static int aic3x_remove(struct snd_soc_codec *codec) aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); list_del(&aic3x->list); - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) { - gpio_set_value(aic3x->gpio_reset, 0); - gpio_free(aic3x->gpio_reset); - } for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); - regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); return 0; } @@ -1484,7 +1455,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_priv *aic3x; struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; - int ret; + int ret, i; u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1545,14 +1516,46 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->model = id->driver_data; + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err; + gpio_direction_output(aic3x->gpio_reset, 0); + } + + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) + aic3x->supplies[i].supply = aic3x_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(aic3x->supplies), + aic3x->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + goto err_gpio; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); return ret; + +err_gpio: + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) + gpio_free(aic3x->gpio_reset); +err: + return ret; } static int aic3x_i2c_remove(struct i2c_client *client) { + struct aic3x_priv *aic3x = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } return 0; } -- cgit From f9df1ae6b59e5bb16d3094e9c1c8b6feeaf32aae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 23:53:16 +0100 Subject: ASoC: tlv320aic3x: Move to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index de17a36beb6f..397a2133e2d1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1369,8 +1369,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - snd_soc_add_codec_controls(codec, aic3x_snd_controls, - ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); @@ -1428,6 +1426,8 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .remove = aic3x_remove, .suspend = aic3x_suspend, .resume = aic3x_resume, + .controls = aic3x_snd_controls, + .num_controls = ARRAY_SIZE(aic3x_snd_controls), }; /* -- cgit From 58a63fbd7c80510140a94442b2ca9199bb6d51c3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 23:57:36 +0100 Subject: ASoC: tlv320aic3x: Move to table based DAPM init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 397a2133e2d1..16fc74cae754 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -818,12 +818,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - if (aic3x->model == AIC3X_MODEL_3007) { snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); @@ -1428,6 +1422,10 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .resume = aic3x_resume, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; /* -- cgit From 2677b4bb7316c07dd53535e01bd9b2ec699d0314 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:55:39 +0100 Subject: ASoC: tlv320aic3x: Don't reference cache datastructure directly Rather than referencing the cache directly read back the values we are going to restore, supporting refactoring to use regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 16fc74cae754..83e7d855c49a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1068,14 +1068,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int aic3x_init_3007(struct snd_soc_codec *codec) { - u8 tmp1, tmp2, *cache = codec->reg_cache; + unsigned int tmp1, tmp2; /* * There is no need to cache writes to undocumented page 0xD but * respective page 0 register cache entries must be preserved */ - tmp1 = cache[0xD]; - tmp2 = cache[0x8]; + tmp1 = snd_soc_read(codec, 0xD); + tmp2 = snd_soc_read(codec, 0x8); /* Class-D speaker driver init; datasheet p. 46 */ snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); snd_soc_write(codec, 0xD, 0x0D); @@ -1083,8 +1083,9 @@ static int aic3x_init_3007(struct snd_soc_codec *codec) snd_soc_write(codec, 0x8, 0x5D); snd_soc_write(codec, 0x8, 0x5C); snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); - cache[0xD] = tmp1; - cache[0x8] = tmp2; + + snd_soc_write(codec, 0xD, tmp1); + snd_soc_write(codec, 0x8, tmp2); return 0; } -- cgit From 2a6fedec195b9bd20e60f9825ba7cc6315e54652 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 00:07:13 +0100 Subject: ASoC: tlv320aic3x: Convert to direct regmap API usage This is slightly more complex than a standard regmap conversion due to the moderately detailed cache control and the open coding of a register patch for the class D speaker on the TLV320AIC3007. Although the device supports paging this is not currently implemented as the additional pages are only used during the application of the patch for the TLV320AIC3007. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 151 ++++++++++++++++++++--------------------- 1 file changed, 73 insertions(+), 78 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 83e7d855c49a..892c108ca67a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -72,9 +72,9 @@ struct aic3x_disable_nb { /* codec private data */ struct aic3x_priv { struct snd_soc_codec *codec; + struct regmap *regmap; struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; - enum snd_soc_control_type control_type; struct aic3x_setup_data *setup; unsigned int sysclk; struct list_head list; @@ -90,35 +90,45 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, 0x00, /* 100 */ - 0x00, 0x00, 0x00, 0x00, /* 104 */ - 0x00, 0x00, /* 108 */ +static const struct reg_default aic3x_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x10 }, + { 4, 0x04 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x01 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x80 }, + { 16, 0x80 }, { 17, 0xff }, { 18, 0xff }, { 19, 0x78 }, + { 20, 0x78 }, { 21, 0x78 }, { 22, 0x78 }, { 23, 0x78 }, + { 24, 0x78 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0xfe }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0xfe }, { 31, 0x00 }, + { 32, 0x18 }, { 33, 0x18 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x00 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x80 }, + { 44, 0x80 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x04 }, + { 52, 0x00 }, { 53, 0x00 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x04 }, { 59, 0x00 }, + { 60, 0x00 }, { 61, 0x00 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x04 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x04 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x00 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x00 }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x00 }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x02 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, +}; + +static const struct regmap_config aic3x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DAC_ICC_ADJ, + .reg_defaults = aic3x_reg, + .num_reg_defaults = ARRAY_SIZE(aic3x_reg), + .cache_type = REGCACHE_RBTREE, }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ @@ -1066,30 +1076,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int aic3x_init_3007(struct snd_soc_codec *codec) -{ - unsigned int tmp1, tmp2; - - /* - * There is no need to cache writes to undocumented page 0xD but - * respective page 0 register cache entries must be preserved - */ - tmp1 = snd_soc_read(codec, 0xD); - tmp2 = snd_soc_read(codec, 0x8); - /* Class-D speaker driver init; datasheet p. 46 */ - snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); - snd_soc_write(codec, 0xD, 0x0D); - snd_soc_write(codec, 0x8, 0x5C); - snd_soc_write(codec, 0x8, 0x5D); - snd_soc_write(codec, 0x8, 0x5C); - snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); - - snd_soc_write(codec, 0xD, tmp1); - snd_soc_write(codec, 0x8, tmp2); - - return 0; -} - static int aic3x_regulator_event(struct notifier_block *nb, unsigned long event, void *data) { @@ -1104,7 +1090,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, */ if (gpio_is_valid(aic3x->gpio_reset)) gpio_set_value(aic3x->gpio_reset, 0); - aic3x->codec->cache_sync = 1; + regcache_mark_dirty(aic3x->regmap); } return 0; @@ -1113,8 +1099,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, static int aic3x_set_power(struct snd_soc_codec *codec, int power) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - int i, ret; - u8 *cache = codec->reg_cache; + int ret; if (power) { ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies), @@ -1122,12 +1107,6 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) if (ret) goto out; aic3x->power = 1; - /* - * Reset release and cache sync is necessary only if some - * supply was off or if there were cached writes - */ - if (!codec->cache_sync) - goto out; if (gpio_is_valid(aic3x->gpio_reset)) { udelay(1); @@ -1135,12 +1114,8 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) } /* Sync reg_cache with the hardware */ - codec->cache_only = 0; - for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) - snd_soc_write(codec, i, cache[i]); - if (aic3x->model == AIC3X_MODEL_3007) - aic3x_init_3007(codec); - codec->cache_sync = 0; + regcache_cache_only(aic3x->regmap, false); + regcache_sync(aic3x->regmap); } else { /* * Do soft reset to this codec instance in order to clear @@ -1148,10 +1123,10 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) * remain on */ snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); - codec->cache_sync = 1; + regcache_mark_dirty(aic3x->regmap); aic3x->power = 0; /* HW writes are needless when bias is off */ - codec->cache_only = 1; + regcache_cache_only(aic3x->regmap, true); ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); } @@ -1306,7 +1281,6 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { - aic3x_init_3007(codec); snd_soc_write(codec, CLASSD_CTRL, 0); } @@ -1334,7 +1308,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + 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; @@ -1353,7 +1327,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) } } - codec->cache_only = 1; + regcache_mark_dirty(aic3x->regmap); aic3x_init(codec); if (aic3x->setup) { @@ -1414,9 +1388,6 @@ static int aic3x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(aic3x_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = aic3x_reg, .probe = aic3x_probe, .remove = aic3x_remove, .suspend = aic3x_suspend, @@ -1443,6 +1414,16 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); +static const struct reg_default aic3007_class_d[] = { + /* Class-D speaker driver init; datasheet p. 46 */ + { AIC3X_PAGE_SELECT, 0x0D }, + { 0xD, 0x0D }, + { 0x8, 0x5C }, + { 0x8, 0x5D }, + { 0x8, 0x5C }, + { AIC3X_PAGE_SELECT, 0x00 }, +}; + /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -1463,7 +1444,13 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } - aic3x->control_type = SND_SOC_I2C; + aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap); + if (IS_ERR(aic3x->regmap)) { + ret = PTR_ERR(aic3x->regmap); + return ret; + } + + regcache_cache_only(aic3x->regmap, true); i2c_set_clientdata(i2c, aic3x); if (pdata) { @@ -1533,6 +1520,14 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + if (aic3x->model == AIC3X_MODEL_3007) { + ret = regmap_register_patch(aic3x->regmap, aic3007_class_d, + ARRAY_SIZE(aic3007_class_d)); + if (ret != 0) + dev_err(&i2c->dev, "Failed to init class D: %d\n", + ret); + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); return ret; -- cgit From 19ab2a7a24539d6c80dfe301d2970b075ad3b9ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 11:14:39 +0100 Subject: ASoC: max98088: Set max_register Makes some of the debug functions more useful. Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 31912d59702c..66ceee22fdad 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -552,6 +552,7 @@ static const struct regmap_config max98088_regmap = { .readable_reg = max98088_readable_register, .volatile_reg = max98088_volatile_register, + .max_register = 0xff, .reg_defaults = max98088_reg, .num_reg_defaults = ARRAY_SIZE(max98088_reg), -- cgit From 068416620c0d956b3b382d19dd3000119e280f8c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 18:12:51 +0100 Subject: ASoC: max9850: Convert to direct regmap API usage This prepares for removal of the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 58c38a5b481c..c5dd61785f8d 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -27,18 +28,26 @@ #include "max9850.h" struct max9850_priv { + struct regmap *regmap; unsigned int sysclk; }; /* max9850 register cache */ -static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { - 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +static const struct reg_default max9850_reg[] = { + { 2, 0x0c }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, }; /* these registers are not used at the moment but provided for the sake of * completeness */ -static int max9850_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool max9850_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case MAX9850_STATUSA: @@ -49,6 +58,15 @@ static int max9850_volatile_register(struct snd_soc_codec *codec, } } +static const struct regmap_config max9850_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX9850_DIGITAL_AUDIO, + .volatile_reg = max9850_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), @@ -225,6 +243,7 @@ static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int max9850_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -234,7 +253,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max9850->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -295,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + 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; @@ -316,10 +335,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .suspend = max9850_suspend, .resume = max9850_resume, .set_bias_level = max9850_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max9850_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max9850_reg, - .volatile_register = max9850_volatile_register, .controls = max9850_controls, .num_controls = ARRAY_SIZE(max9850_controls), @@ -340,6 +355,10 @@ static int max9850_i2c_probe(struct i2c_client *i2c, if (max9850 == NULL) return -ENOMEM; + max9850->regmap = devm_regmap_init_i2c(i2c, &max9850_regmap); + if (IS_ERR(max9850->regmap)) + return PTR_ERR(max9850->regmap); + i2c_set_clientdata(i2c, max9850); ret = snd_soc_register_codec(&i2c->dev, -- cgit From 6f88063c1474b8cdd9254d3934047b6087222145 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:46:05 +0100 Subject: ASoC: cq93vc: Use table based control registration Saves a little code. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index e2c4c0a896e2..e538f4eca980 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -156,10 +156,6 @@ static int cq93vc_probe(struct snd_soc_codec *codec) davinci_vc->cq93vc.codec = codec; codec->control_data = davinci_vc; - /* Set controls */ - snd_soc_add_codec_controls(codec, cq93vc_snd_controls, - ARRAY_SIZE(cq93vc_snd_controls)); - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -180,6 +176,8 @@ static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .probe = cq93vc_probe, .remove = cq93vc_remove, .resume = cq93vc_resume, + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }; static int cq93vc_platform_probe(struct platform_device *pdev) -- cgit From 1201939a6f981cb656872784e39ef443540078cd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:38:16 +0100 Subject: ASoC: cq93vc: Use core I/O functions Support future refactoring by using the core I/O functions rather than calling the driver provided I/O functions directly. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index e538f4eca980..2cbb584b33de 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -64,13 +64,15 @@ static const struct snd_kcontrol_new cq93vc_snd_controls[] = { static int cq93vc_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + u8 reg; if (mute) - cq93vc_write(codec, DAVINCI_VC_REG09, - reg | DAVINCI_VC_REG09_MUTE); + reg = DAVINCI_VC_REG09_MUTE; else - cq93vc_write(codec, DAVINCI_VC_REG09, reg); + reg = 0; + + snd_soc_update_bits(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_MUTE, + reg); return 0; } @@ -97,18 +99,18 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_ON: - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_ON); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_OFF); break; case SND_SOC_BIAS_OFF: /* force all power off */ - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_OFF); break; } -- cgit From d33c33352b3228ca3a422e55981f80fc12dc30f8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 13:52:41 +0100 Subject: ASoC: cq93vc: Use regmap for I/O Avoid use of the ASoC-specific register I/O functions by converting to use the MMIO regmap provided the core MFD. Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 24 +++--------------------- 1 file changed, 3 insertions(+), 21 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 2cbb584b33de..43737a27d79c 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -38,24 +38,6 @@ #include #include -static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct davinci_vc *davinci_vc = codec->control_data; - - return readl(davinci_vc->base + reg); -} - -static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct davinci_vc *davinci_vc = codec->control_data; - - writel(value, davinci_vc->base + reg); - - return 0; -} - static const struct snd_kcontrol_new cq93vc_snd_controls[] = { SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), @@ -156,7 +138,9 @@ static int cq93vc_probe(struct snd_soc_codec *codec) struct davinci_vc *davinci_vc = codec->dev->platform_data; davinci_vc->cq93vc.codec = codec; - codec->control_data = davinci_vc; + codec->control_data = davinci_vc->regmap; + + snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -172,8 +156,6 @@ static int cq93vc_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { - .read = cq93vc_read, - .write = cq93vc_write, .set_bias_level = cq93vc_set_bias_level, .probe = cq93vc_probe, .remove = cq93vc_remove, -- cgit From 4aa11d67b66a84189d25f301e7ef206c4f541692 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Sep 2013 19:26:08 +0100 Subject: ASoC: tlv320aic23: Convert to direct regmap API usage This moves us towards being able to remove the duplicated register I/O code in ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 54 ++++++++++++++++++++++++------------------ 1 file changed, 31 insertions(+), 23 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 3a6be8c3d557..5d430cc56f51 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -37,11 +38,27 @@ /* * AIC23 register cache */ -static const u16 tlv320aic23_reg[] = { - 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ - 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ - 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +static const struct reg_default tlv320aic23_reg[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x00F9 }, + { 3, 0x00F9 }, + { 4, 0x001A }, + { 5, 0x0004 }, + { 6, 0x0007 }, + { 7, 0x0001 }, + { 8, 0x0020 }, + { 9, 0x0000 }, +}; + +static const struct regmap_config tlv320aic23_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = TLV320AIC23_RESET, + .reg_defaults = tlv320aic23_reg, + .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), + .cache_type = REGCACHE_RBTREE, }; static const char *rec_src_text[] = { "Line", "Mic" }; @@ -171,7 +188,7 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { - enum snd_soc_control_type control_type; + struct regmap *regmap; int mclk; int requested_adc; int requested_dac; @@ -532,7 +549,9 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec) static int tlv320aic23_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); + regcache_mark_dirty(aic23->regmap); + regcache_sync(aic23->regmap); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -540,10 +559,9 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { - struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -552,16 +570,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); - /* Write the register default value to cache for reserved registers, - * so the write to the these registers are suppressed by the cache - * restore code when it skips writes of default registers. - */ - snd_soc_cache_write(codec, 0x0A, 0); - snd_soc_cache_write(codec, 0x0B, 0); - snd_soc_cache_write(codec, 0x0C, 0); - snd_soc_cache_write(codec, 0x0D, 0); - snd_soc_cache_write(codec, 0x0E, 0); - /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -596,9 +604,6 @@ static int tlv320aic23_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { - .reg_cache_size = ARRAY_SIZE(tlv320aic23_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = tlv320aic23_reg, .probe = tlv320aic23_probe, .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, @@ -629,8 +634,11 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, if (aic23 == NULL) return -ENOMEM; + aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); + if (IS_ERR(aic23->regmap)) + return PTR_ERR(aic23->regmap); + i2c_set_clientdata(i2c, aic23); - aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); -- cgit From 806955dd9cf071ecd99acbaa8c73ae1f34dcf83d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 13:10:33 +0100 Subject: ASoC: tlv320aic26: Convert to table based control init Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 7b8f3d965f43..32c6b0768e56 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -377,7 +377,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, err, i, reg; + int ret, i, reg; aic26->codec = codec; @@ -403,12 +403,6 @@ static int aic26_probe(struct snd_soc_codec *codec) if (ret) dev_info(codec->dev, "error creating sysfs files\n"); - /* register controls */ - dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_codec_controls(codec, aic26_snd_controls, - ARRAY_SIZE(aic26_snd_controls)); - WARN_ON(err < 0); - return 0; } @@ -418,6 +412,8 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .controls = aic26_snd_controls, + .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), .dapm_routes = tlv320aic26_dapm_routes, -- cgit From 5b0959d472c215e6d712ac47e64110bd125ddd07 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 13:14:41 +0100 Subject: ASoC: tlv320aic26: Use snd_soc_update_bits() Use snd_soc_update_bits() rather than open coding. Since the register cache is currently only used where update_bits() is used this means the current register cache can be removed entirely. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 51 +++++++++++------------------------------- 1 file changed, 13 insertions(+), 38 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 32c6b0768e56..4d8244750f23 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -74,19 +74,6 @@ static unsigned int aic26_reg_read(struct snd_soc_codec *codec, return value; } -static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return 0; - } - - return cache[reg]; -} - static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -195,19 +182,15 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); - reg &= ~0xf800; if (aic26->master) - reg |= 0x0800; + reg = 0x0800; if (fsref == 48000) - reg |= 0x2000; - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + reg = 0x2000; + snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL3, 0xf800, reg); /* Audio Control 1 (FSref divisor) */ - reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); - reg &= ~0x0fff; - reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + reg = wlen | aic26->datfm | (divisor << 3) | divisor; + snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL1, 0xfff, reg); return 0; } @@ -219,16 +202,16 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN); + u16 reg; dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n", dai, mute); if (mute) - reg |= 0x8080; + reg = 0x8080; else - reg &= ~0x8080; - snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); + reg = 0; + snd_soc_update_bits(codec, AIC26_REG_DAC_GAIN, 0x8000, reg); return 0; } @@ -346,7 +329,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = snd_soc_read(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -360,11 +343,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, const char *buf, size_t count) { struct aic26 *aic26 = dev_get_drvdata(dev); - int val; - val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); - val |= 0x8000; - snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_update_bits(aic26->codec, AIC26_REG_AUDIO_CTRL2, + 0x8000, 0x800); return count; } @@ -377,7 +358,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, i, reg; + int ret, reg; aic26->codec = codec; @@ -393,10 +374,6 @@ static int aic26_probe(struct snd_soc_codec *codec) reg |= 0x0800; /* set master mode */ snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); - /* Fill register cache */ - for (i = 0; i < codec->driver->reg_cache_size; i++) - snd_soc_read(codec, i); - /* Register the sysfs files for debugging */ /* Create SysFS files */ ret = device_create_file(codec->dev, &dev_attr_keyclick); @@ -410,8 +387,6 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .probe = aic26_probe, .read = aic26_reg_read, .write = aic26_reg_write, - .reg_cache_size = AIC26_NUM_REGS, - .reg_word_size = sizeof(u16), .controls = aic26_snd_controls, .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, -- cgit From 7fbdeb809050cb958f3baa83dcc643f9a2f287f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 13:29:44 +0100 Subject: ASoC: tlv320aic26: Convert to direct regmap API usage This moves us towards being able to remove the duplicated register I/O code in ASoC. The datasheet and the driver document the device as having a register map divided into pages but since the paging is actually done by sending the page address and the register address with each transaction this is no different to having a simple register address. The datasheet does also document the low five bits of the 16 bit "command" as unused which we could represent as padding but it seems simpler and less confusing to things that use block transfers or autoincrement to represent these as part of the register address. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 80 +++++++----------------------------------- sound/soc/codecs/tlv320aic26.h | 5 +-- 2 files changed, 13 insertions(+), 72 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 4d8244750f23..94a658fa6d97 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,6 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; + struct regmap *regmap; struct snd_soc_codec *codec; int master; int datfm; @@ -40,72 +41,6 @@ struct aic26 { int keyclick_len; }; -/* --------------------------------------------------------------------- - * Register access routines - */ -static unsigned int aic26_reg_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - u16 cmd, value; - u8 buffer[2]; - int rc; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return 0; - } - - /* Do SPI transfer; first 16bits are command; remaining is - * register contents */ - cmd = AIC26_READ_COMMAND_WORD(reg); - buffer[0] = (cmd >> 8) & 0xff; - buffer[1] = cmd & 0xff; - rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2); - if (rc) { - dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); - return -EIO; - } - value = (buffer[0] << 8) | buffer[1]; - - /* Update the cache before returning with the value */ - cache[reg] = value; - return value; -} - -static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - u16 cmd; - u8 buffer[4]; - int rc; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return -EINVAL; - } - - /* Do SPI transfer; first 16bits are command; remaining is data - * to write into register */ - cmd = AIC26_WRITE_COMMAND_WORD(reg); - buffer[0] = (cmd >> 8) & 0xff; - buffer[1] = cmd & 0xff; - buffer[2] = value >> 8; - buffer[3] = value; - rc = spi_write(aic26->spi, buffer, 4); - if (rc) { - dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); - return -EIO; - } - - /* update cache before returning */ - cache[reg] = value; - return 0; -} - static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICIN"), SND_SOC_DAPM_INPUT("AUX"), @@ -360,6 +295,8 @@ static int aic26_probe(struct snd_soc_codec *codec) struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, reg; + snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + aic26->codec = codec; /* Reset the codec to power on defaults */ @@ -385,8 +322,6 @@ static int aic26_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver aic26_soc_codec_dev = { .probe = aic26_probe, - .read = aic26_reg_read, - .write = aic26_reg_write, .controls = aic26_snd_controls, .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, @@ -395,6 +330,11 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), }; +static const struct regmap_config aic26_regmap = { + .reg_bits = 16, + .val_bits = 16, +}; + /* --------------------------------------------------------------------- * SPI device portion of driver: probe and release routines and SPI * driver registration. @@ -411,6 +351,10 @@ static int aic26_spi_probe(struct spi_device *spi) if (!aic26) return -ENOMEM; + aic26->regmap = devm_regmap_init_spi(spi, &aic26_regmap); + if (IS_ERR(aic26->regmap)) + return PTR_ERR(aic26->regmap); + /* Initialize the driver data */ aic26->spi = spi; dev_set_drvdata(&spi->dev, aic26); diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 67f19c3bebe6..629b85e75409 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -9,10 +9,7 @@ #define _TLV320AIC16_H_ /* AIC26 Registers */ -#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5)) -#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5)) -#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) -#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) +#define AIC26_PAGE_ADDR(page, offset) ((page << 11) | offset << 5) /* Page 0: Auxiliary data registers */ #define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) -- cgit From fd792f8fbcfa95674b6c417429f576ad1d808086 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Sep 2013 19:14:32 +0100 Subject: mfd: mc13xxx: Move SPI erratum workaround into SPI I/O function Move the workaround for double sending AUDIO_CODEC and AUDIO_DAC writes into the SPI core, aiding refactoring to eliminate the ASoC custom I/O functions and avoiding the extra writes for I2C. Signed-off-by: Mark Brown Signed-off-by: Lee Jones --- sound/soc/codecs/mc13783.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index ea141e1d6f28..4d3c8fd8c5db 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -125,10 +125,6 @@ static int mc13783_write(struct snd_soc_codec *codec, ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - /* include errata fix for spi audio problems */ - if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) - ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - mc13xxx_unlock(priv->mc13xxx); return ret; -- cgit From 2d9215c1ecd6f133952bc081a288dbb180816290 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Sep 2013 19:04:17 +0100 Subject: ASoC: mc13783: Use regmap directly from ASoC As part of a push to remove the register I/O functionality from ASoC (since it is now duplicated in the regmap API) convert the mc13783 driver to use regmap directly. Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 55 ++++++++-------------------------------------- 1 file changed, 9 insertions(+), 46 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 4d3c8fd8c5db..eedbf05b8e96 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -30,16 +30,10 @@ #include #include #include +#include #include "mc13783.h" -#define MC13783_AUDIO_RX0 36 -#define MC13783_AUDIO_RX1 37 -#define MC13783_AUDIO_TX 38 -#define MC13783_SSI_NETWORK 39 -#define MC13783_AUDIO_CODEC 40 -#define MC13783_AUDIO_DAC 41 - #define AUDIO_RX0_ALSPEN (1 << 5) #define AUDIO_RX0_ALSPSEL (1 << 7) #define AUDIO_RX0_ADDCDC (1 << 21) @@ -95,41 +89,12 @@ struct mc13783_priv { struct mc13xxx *mc13xxx; + struct regmap *regmap; enum mc13783_ssi_port adc_ssi_port; enum mc13783_ssi_port dac_ssi_port; }; -static unsigned int mc13783_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - unsigned int value = 0; - - mc13xxx_lock(priv->mc13xxx); - - mc13xxx_reg_read(priv->mc13xxx, reg, &value); - - mc13xxx_unlock(priv->mc13xxx); - - return value; -} - -static int mc13783_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; - - mc13xxx_lock(priv->mc13xxx); - - ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - - mc13xxx_unlock(priv->mc13xxx); - - return ret; -} - /* Mapping between sample rates and register value */ static unsigned int mc13783_rates[] = { 8000, 11025, 12000, 16000, @@ -583,8 +548,14 @@ static struct snd_kcontrol_new mc13783_control_list[] = { static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; - mc13xxx_lock(priv->mc13xxx); + codec->control_data = dev_get_regmap(codec->dev->parent, NULL); + ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* these are the reset values */ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); @@ -608,8 +579,6 @@ static int mc13783_probe(struct snd_soc_codec *codec) mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, 0, AUDIO_SSI_SEL); - mc13xxx_unlock(priv->mc13xxx); - return 0; } @@ -617,13 +586,9 @@ static int mc13783_remove(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - mc13xxx_lock(priv->mc13xxx); - /* Make sure VAUDIOON is off */ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0); - mc13xxx_unlock(priv->mc13xxx); - return 0; } @@ -713,8 +678,6 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { .probe = mc13783_probe, .remove = mc13783_remove, - .read = mc13783_read, - .write = mc13783_write, .controls = mc13783_control_list, .num_controls = ARRAY_SIZE(mc13783_control_list), .dapm_widgets = mc13783_dapm_widgets, -- cgit From 83cbe35b874621a23ca468621c0d833b76a1b8de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 19:22:54 +0100 Subject: ASoC: sn95031: Convert to regmap This moves us towards being able to remove the duplicated register I/O functionality in ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index dba26e63844e..13045f2af4d3 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -164,30 +164,28 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) } /*end - adc helper functions */ -static inline unsigned int sn95031_read(struct snd_soc_codec *codec, - unsigned int reg) +static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val) { u8 value = 0; int ret; ret = intel_scu_ipc_ioread8(reg, &value); - if (ret) - pr_err("read of %x failed, err %d\n", reg, ret); - return value; + if (ret == 0) + *val = value; + return ret; } -static inline int sn95031_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int sn95031_write(void *ctx, unsigned int reg, unsigned int value) { - int ret; - - ret = intel_scu_ipc_iowrite8(reg, value); - if (ret) - pr_err("write of %x failed, err %d\n", reg, ret); - return ret; + return intel_scu_ipc_iowrite8(reg, value); } +static const struct regmap_config sn95031_regmap = { + .reg_read = sn95031_read, + .reg_write = sn95031_write, +}; + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -827,6 +825,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -886,8 +886,6 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, @@ -898,7 +896,14 @@ static struct snd_soc_codec_driver sn95031_codec = { static int sn95031_device_probe(struct platform_device *pdev) { + struct regmap *regmap; + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + + regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } -- cgit From 752b776435cb35da27a0bbec8deecc33b3461288 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 11:36:26 +0100 Subject: ASoC: tlv320aic32x4: Move GPIO acquisition to I2C probe This is more idiomatic and interacts better with deferred probe. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2ed57d4aa445..cf70bf86c344 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -617,16 +617,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - int ret; codec->hw_write = (hw_write_t) i2c_master_send; codec->control_data = aic32x4->control_data; if (aic32x4->rstn_gpio >= 0) { - ret = devm_gpio_request_one(codec->dev, aic32x4->rstn_gpio, - GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); - if (ret != 0) - return ret; ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -735,6 +730,13 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } + if (aic32x4->rstn_gpio >= 0) { + ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); return ret; -- cgit From 4d208ca429ad424595fd08c0cca323605ebfc38b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Sep 2013 11:37:53 +0100 Subject: ASoC: tlv320aic32x4: Convert to direct regmap API usage This moves us towards being able to remove the duplicate register I/O functionality in ASoC and saves some code. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 89 +++++++++++----------------------------- 1 file changed, 23 insertions(+), 66 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index cf70bf86c344..18cdcca9014c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,9 +60,8 @@ struct aic32x4_rate_divs { }; struct aic32x4_priv { + struct regmap *regmap; u32 sysclk; - u8 page_no; - void *control_data; u32 power_cfg; u32 micpga_routing; bool swapdacs; @@ -262,67 +261,25 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"Right ADC", NULL, "Right Input Mixer"}, }; -static inline int aic32x4_change_page(struct snd_soc_codec *codec, - unsigned int new_page) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 data[2]; - int ret; - - data[0] = 0x00; - data[1] = new_page & 0xff; - - ret = codec->hw_write(codec->control_data, data, 2); - if (ret == 2) { - aic32x4->page_no = new_page; - return 0; - } else { - return ret; - } -} - -static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - unsigned int page = reg / 128; - unsigned int fixed_reg = reg % 128; - u8 data[2]; - int ret; - - /* A write to AIC32X4_PSEL is really a non-explicit page change */ - if (reg == AIC32X4_PSEL) - return aic32x4_change_page(codec, val); - - if (aic32x4->page_no != page) { - ret = aic32x4_change_page(codec, page); - if (ret != 0) - return ret; - } +static const struct regmap_range_cfg aic32x4_regmap_pages[] = { + { + .selector_reg = 0, + .selector_mask = 0xff, + .window_start = 0, + .window_len = 128, + .range_min = AIC32X4_PAGE1, + .range_max = AIC32X4_PAGE1 + 127, + }, +}; - data[0] = fixed_reg & 0xff; - data[1] = val & 0xff; +static const struct regmap_config aic32x4_regmap = { + .reg_bits = 8, + .val_bits = 8, - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - unsigned int page = reg / 128; - unsigned int fixed_reg = reg % 128; - int ret; - - if (aic32x4->page_no != page) { - ret = aic32x4_change_page(codec, page); - if (ret != 0) - return ret; - } - return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); -} + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; static inline int aic32x4_get_divs(int mclk, int rate) { @@ -618,8 +575,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - codec->hw_write = (hw_write_t) i2c_master_send; - codec->control_data = aic32x4->control_data; + snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (aic32x4->rstn_gpio >= 0) { ndelay(10); @@ -687,8 +643,6 @@ static int aic32x4_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { - .read = aic32x4_read, - .write = aic32x4_write, .probe = aic32x4_probe, .remove = aic32x4_remove, .suspend = aic32x4_suspend, @@ -715,7 +669,10 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, if (aic32x4 == NULL) return -ENOMEM; - aic32x4->control_data = i2c; + aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); + if (IS_ERR(aic32x4->regmap)) + return PTR_ERR(aic32x4->regmap); + i2c_set_clientdata(i2c, aic32x4); if (pdata) { -- cgit From d6173df35f2dbd0e11f2361fc979ebf2e53cb6cc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Sep 2013 19:36:11 +0100 Subject: ASoC: si476x: Remove custom register I/O implementation The current si476x I/O implementation wraps the regmap for the core with functions that make the register map cache only when the device is powered down. This implementation appears to be incomplete since there is no code to synchronise the cache so writes done while the core is powered down will be ignored, the device will only be configured if it is powered. A better and more idiomatic approach would be to have the MFD manage the cache, making the device cache only when it powers things down. This also allows ASoC to use the standard regmap helpers for the device which helps remove the ASoC custom ones so do convert to do that. Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 46 +--------------------------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38f3b105c17d..03645ce42063 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -60,48 +60,6 @@ enum si476x_pcm_format { SI476X_PCM_FORMAT_S24_LE = 6, }; -static unsigned int si476x_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - int err; - unsigned int val; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_read(core->regmap, reg, &val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - if (err < 0) - return err; - - return val; -} - -static int si476x_codec_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val) -{ - int err; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_write(core->regmap, reg, val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - return err; -} - static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), @@ -239,7 +197,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = i2c_mfd_cell_to_core(codec->dev); + codec->control_data = dev_get_regmap(codec->dev->parent, NULL); return 0; } @@ -268,8 +226,6 @@ static struct snd_soc_dai_driver si476x_dai = { static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, - .read = si476x_codec_read, - .write = si476x_codec_write, .dapm_widgets = si476x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), .dapm_routes = si476x_dapm_routes, -- cgit From c3df37c9380d70f19a9cb2de4c7d58d7822a4b35 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 13:47:08 +0200 Subject: ASoC: adau1373: Convert to direct regmap usage Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 295 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 235 insertions(+), 60 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1aa10ddf3a61..c57c1f81a611 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -32,6 +32,7 @@ struct adau1373_dai { }; struct adau1373 { + struct regmap *regmap; struct adau1373_dai dais[3]; }; @@ -152,37 +153,172 @@ struct adau1373 { #define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 #define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 -static const uint8_t adau1373_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ - 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ - 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ - 0x00, 0x1f, 0x0f, 0x00, 0x00, +static const struct reg_default adau1373_reg_defaults[] = { + { ADAU1373_INPUT_MODE, 0x00 }, + { ADAU1373_AINL_CTRL(0), 0x00 }, + { ADAU1373_AINR_CTRL(0), 0x00 }, + { ADAU1373_AINL_CTRL(1), 0x00 }, + { ADAU1373_AINR_CTRL(1), 0x00 }, + { ADAU1373_AINL_CTRL(2), 0x00 }, + { ADAU1373_AINR_CTRL(2), 0x00 }, + { ADAU1373_AINL_CTRL(3), 0x00 }, + { ADAU1373_AINR_CTRL(3), 0x00 }, + { ADAU1373_LLINE_OUT(0), 0x00 }, + { ADAU1373_RLINE_OUT(0), 0x00 }, + { ADAU1373_LLINE_OUT(1), 0x00 }, + { ADAU1373_RLINE_OUT(1), 0x00 }, + { ADAU1373_LSPK_OUT, 0x00 }, + { ADAU1373_RSPK_OUT, 0x00 }, + { ADAU1373_LHP_OUT, 0x00 }, + { ADAU1373_RHP_OUT, 0x00 }, + { ADAU1373_ADC_GAIN, 0x00 }, + { ADAU1373_LADC_MIXER, 0x00 }, + { ADAU1373_RADC_MIXER, 0x00 }, + { ADAU1373_LLINE1_MIX, 0x00 }, + { ADAU1373_RLINE1_MIX, 0x00 }, + { ADAU1373_LLINE2_MIX, 0x00 }, + { ADAU1373_RLINE2_MIX, 0x00 }, + { ADAU1373_LSPK_MIX, 0x00 }, + { ADAU1373_RSPK_MIX, 0x00 }, + { ADAU1373_LHP_MIX, 0x00 }, + { ADAU1373_RHP_MIX, 0x00 }, + { ADAU1373_EP_MIX, 0x00 }, + { ADAU1373_HP_CTRL, 0x00 }, + { ADAU1373_HP_CTRL2, 0x00 }, + { ADAU1373_LS_CTRL, 0x00 }, + { ADAU1373_EP_CTRL, 0x00 }, + { ADAU1373_MICBIAS_CTRL1, 0x00 }, + { ADAU1373_MICBIAS_CTRL2, 0x00 }, + { ADAU1373_OUTPUT_CTRL, 0x00 }, + { ADAU1373_PWDN_CTRL1, 0x00 }, + { ADAU1373_PWDN_CTRL2, 0x00 }, + { ADAU1373_PWDN_CTRL3, 0x00 }, + { ADAU1373_DPLL_CTRL(0), 0x00 }, + { ADAU1373_PLL_CTRL1(0), 0x00 }, + { ADAU1373_PLL_CTRL2(0), 0x00 }, + { ADAU1373_PLL_CTRL3(0), 0x00 }, + { ADAU1373_PLL_CTRL4(0), 0x00 }, + { ADAU1373_PLL_CTRL5(0), 0x00 }, + { ADAU1373_PLL_CTRL6(0), 0x02 }, + { ADAU1373_DPLL_CTRL(1), 0x00 }, + { ADAU1373_PLL_CTRL1(1), 0x00 }, + { ADAU1373_PLL_CTRL2(1), 0x00 }, + { ADAU1373_PLL_CTRL3(1), 0x00 }, + { ADAU1373_PLL_CTRL4(1), 0x00 }, + { ADAU1373_PLL_CTRL5(1), 0x00 }, + { ADAU1373_PLL_CTRL6(1), 0x02 }, + { ADAU1373_HEADDECT, 0x00 }, + { ADAU1373_ADC_CTRL, 0x00 }, + { ADAU1373_CLK_SRC_DIV(0), 0x00 }, + { ADAU1373_CLK_SRC_DIV(1), 0x00 }, + { ADAU1373_DAI(0), 0x0a }, + { ADAU1373_DAI(1), 0x0a }, + { ADAU1373_DAI(2), 0x0a }, + { ADAU1373_BCLKDIV(0), 0x00 }, + { ADAU1373_BCLKDIV(1), 0x00 }, + { ADAU1373_BCLKDIV(2), 0x00 }, + { ADAU1373_SRC_RATIOA(0), 0x00 }, + { ADAU1373_SRC_RATIOB(0), 0x00 }, + { ADAU1373_SRC_RATIOA(1), 0x00 }, + { ADAU1373_SRC_RATIOB(1), 0x00 }, + { ADAU1373_SRC_RATIOA(2), 0x00 }, + { ADAU1373_SRC_RATIOB(2), 0x00 }, + { ADAU1373_DEEMP_CTRL, 0x00 }, + { ADAU1373_SRC_DAI_CTRL(0), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(1), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(2), 0x08 }, + { ADAU1373_DIN_MIX_CTRL(0), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(1), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(2), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(3), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(4), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(0), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(1), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(2), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(3), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(4), 0x00 }, + { ADAU1373_DAI_PBL_VOL(0), 0x00 }, + { ADAU1373_DAI_PBR_VOL(0), 0x00 }, + { ADAU1373_DAI_PBL_VOL(1), 0x00 }, + { ADAU1373_DAI_PBR_VOL(1), 0x00 }, + { ADAU1373_DAI_PBL_VOL(2), 0x00 }, + { ADAU1373_DAI_PBR_VOL(2), 0x00 }, + { ADAU1373_DAI_RECL_VOL(0), 0x00 }, + { ADAU1373_DAI_RECR_VOL(0), 0x00 }, + { ADAU1373_DAI_RECL_VOL(1), 0x00 }, + { ADAU1373_DAI_RECR_VOL(1), 0x00 }, + { ADAU1373_DAI_RECL_VOL(2), 0x00 }, + { ADAU1373_DAI_RECR_VOL(2), 0x00 }, + { ADAU1373_DAC1_PBL_VOL, 0x00 }, + { ADAU1373_DAC1_PBR_VOL, 0x00 }, + { ADAU1373_DAC2_PBL_VOL, 0x00 }, + { ADAU1373_DAC2_PBR_VOL, 0x00 }, + { ADAU1373_ADC_RECL_VOL, 0x00 }, + { ADAU1373_ADC_RECR_VOL, 0x00 }, + { ADAU1373_DMIC_RECL_VOL, 0x00 }, + { ADAU1373_DMIC_RECR_VOL, 0x00 }, + { ADAU1373_VOL_GAIN1, 0x00 }, + { ADAU1373_VOL_GAIN2, 0x00 }, + { ADAU1373_VOL_GAIN3, 0x00 }, + { ADAU1373_HPF_CTRL, 0x00 }, + { ADAU1373_BASS1, 0x00 }, + { ADAU1373_BASS2, 0x00 }, + { ADAU1373_DRC(0) + 0x0, 0x78 }, + { ADAU1373_DRC(0) + 0x1, 0x18 }, + { ADAU1373_DRC(0) + 0x2, 0x00 }, + { ADAU1373_DRC(0) + 0x3, 0x00 }, + { ADAU1373_DRC(0) + 0x4, 0x00 }, + { ADAU1373_DRC(0) + 0x5, 0xc0 }, + { ADAU1373_DRC(0) + 0x6, 0x00 }, + { ADAU1373_DRC(0) + 0x7, 0x00 }, + { ADAU1373_DRC(0) + 0x8, 0x00 }, + { ADAU1373_DRC(0) + 0x9, 0xc0 }, + { ADAU1373_DRC(0) + 0xa, 0x88 }, + { ADAU1373_DRC(0) + 0xb, 0x7a }, + { ADAU1373_DRC(0) + 0xc, 0xdf }, + { ADAU1373_DRC(0) + 0xd, 0x20 }, + { ADAU1373_DRC(0) + 0xe, 0x00 }, + { ADAU1373_DRC(0) + 0xf, 0x00 }, + { ADAU1373_DRC(1) + 0x0, 0x78 }, + { ADAU1373_DRC(1) + 0x1, 0x18 }, + { ADAU1373_DRC(1) + 0x2, 0x00 }, + { ADAU1373_DRC(1) + 0x3, 0x00 }, + { ADAU1373_DRC(1) + 0x4, 0x00 }, + { ADAU1373_DRC(1) + 0x5, 0xc0 }, + { ADAU1373_DRC(1) + 0x6, 0x00 }, + { ADAU1373_DRC(1) + 0x7, 0x00 }, + { ADAU1373_DRC(1) + 0x8, 0x00 }, + { ADAU1373_DRC(1) + 0x9, 0xc0 }, + { ADAU1373_DRC(1) + 0xa, 0x88 }, + { ADAU1373_DRC(1) + 0xb, 0x7a }, + { ADAU1373_DRC(1) + 0xc, 0xdf }, + { ADAU1373_DRC(1) + 0xd, 0x20 }, + { ADAU1373_DRC(1) + 0xe, 0x00 }, + { ADAU1373_DRC(1) + 0xf, 0x00 }, + { ADAU1373_DRC(2) + 0x0, 0x78 }, + { ADAU1373_DRC(2) + 0x1, 0x18 }, + { ADAU1373_DRC(2) + 0x2, 0x00 }, + { ADAU1373_DRC(2) + 0x3, 0x00 }, + { ADAU1373_DRC(2) + 0x4, 0x00 }, + { ADAU1373_DRC(2) + 0x5, 0xc0 }, + { ADAU1373_DRC(2) + 0x6, 0x00 }, + { ADAU1373_DRC(2) + 0x7, 0x00 }, + { ADAU1373_DRC(2) + 0x8, 0x00 }, + { ADAU1373_DRC(2) + 0x9, 0xc0 }, + { ADAU1373_DRC(2) + 0xa, 0x88 }, + { ADAU1373_DRC(2) + 0xb, 0x7a }, + { ADAU1373_DRC(2) + 0xc, 0xdf }, + { ADAU1373_DRC(2) + 0xd, 0x20 }, + { ADAU1373_DRC(2) + 0xe, 0x00 }, + { ADAU1373_DRC(2) + 0xf, 0x00 }, + { ADAU1373_3D_CTRL1, 0x00 }, + { ADAU1373_3D_CTRL2, 0x00 }, + { ADAU1373_FDSP_SEL1, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL4, 0x00 }, + { ADAU1373_DIGMICCTRL, 0x00 }, + { ADAU1373_DIGEN, 0x00 }, }; static const unsigned int adau1373_out_tlv[] = { @@ -418,6 +554,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int pll_id = w->name[3] - '1'; unsigned int val; @@ -426,7 +563,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, else val = 0; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_PLL_EN, val); if (SND_SOC_DAPM_EVENT_ON(event)) @@ -938,7 +1075,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); - snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, (div << 2) | ADAU1373_BCLKDIV_64); @@ -959,7 +1096,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + return regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ADAU1373_DAI_WLEN_MASK, ctrl); } @@ -1016,7 +1153,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ~ADAU1373_DAI_WLEN_MASK, ctrl); return 0; @@ -1039,7 +1176,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, adau1373_dai->sysclk = freq; adau1373_dai->clk_src = clk_id; - snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SOURCE, clk_id << 5); return 0; @@ -1120,6 +1257,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = { static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dpll_div = 0; unsigned int x, r, n, m, i, j, mode; @@ -1187,36 +1325,36 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, if (dpll_div) { dpll_div = 11 - dpll_div; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); } else { - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, ADAU1373_PLL_CTRL6_DPLL_BYPASS); } - snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id), (source << 4) | dpll_div); - snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), (r << 3) | (x << 1) | mode); /* Set sysclk to pll_rate / 4 */ - snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); return 0; } -static void adau1373_load_drc_settings(struct snd_soc_codec *codec, +static void adau1373_load_drc_settings(struct adau1373 *adau1373, unsigned int nr, uint8_t *drc) { unsigned int i; for (i = 0; i < ADAU1373_DRC_SIZE; ++i) - snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); + regmap_write(adau1373->regmap, ADAU1373_DRC(nr) + i, drc[i]); } static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) @@ -1235,13 +1373,14 @@ static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) static int adau1373_probe(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -1256,7 +1395,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) return -EINVAL; for (i = 0; i < pdata->num_drc; ++i) { - adau1373_load_drc_settings(codec, i, + adau1373_load_drc_settings(adau1373, i, pdata->drc_setting[i]); } @@ -1268,18 +1407,18 @@ static int adau1373_probe(struct snd_soc_codec *codec) if (pdata->input_differential[i]) val |= BIT(i); } - snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + regmap_write(adau1373->regmap, ADAU1373_INPUT_MODE, val); val = 0; if (pdata->lineout_differential) val |= ADAU1373_OUTPUT_CTRL_LDIFF; if (pdata->lineout_ground_sense) val |= ADAU1373_OUTPUT_CTRL_LNFBEN; - snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + regmap_write(adau1373->regmap, ADAU1373_OUTPUT_CTRL, val); lineout_differential = pdata->lineout_differential; - snd_soc_write(codec, ADAU1373_EP_CTRL, + regmap_write(adau1373->regmap, ADAU1373_EP_CTRL, (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); } @@ -1289,7 +1428,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) ARRAY_SIZE(adau1373_lineout2_controls)); } - snd_soc_write(codec, ADAU1373_ADC_CTRL, + regmap_write(adau1373->regmap, ADAU1373_ADC_CTRL, ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); return 0; @@ -1298,17 +1437,19 @@ static int adau1373_probe(struct snd_soc_codec *codec) static int adau1373_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, 0); break; } @@ -1324,17 +1465,49 @@ static int adau1373_remove(struct snd_soc_codec *codec) static int adau1373_suspend(struct snd_soc_codec *codec) { - return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adau1373->regmap, true); + + return ret; } static int adau1373_resume(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adau1373->regmap, false); adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_cache_sync(codec); + regcache_sync(adau1373->regmap); return 0; } +static bool adau1373_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1373_SOFT_RESET: + case ADAU1373_ADC_DAC_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config adau1373_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .volatile_reg = adau1373_register_volatile, + .max_register = ADAU1373_SOFT_RESET, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1373_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults), +}; + static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, @@ -1342,9 +1515,6 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), - .reg_cache_default = adau1373_default_regs, - .reg_word_size = sizeof(uint8_t), .set_pll = adau1373_set_pll, @@ -1366,6 +1536,11 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (!adau1373) return -ENOMEM; + adau1373->regmap = devm_regmap_init_i2c(client, + &adau1373_regmap_config); + if (IS_ERR(adau1373->regmap)) + return PTR_ERR(adau1373->regmap); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, -- cgit From 6fb04138a3068609fa0ef3a98b60e31e686b3160 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 13:47:09 +0200 Subject: ASoC: adau1373: Remove ADAU1373_PLL_CTRL7 register definition There is no such register. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index c57c1f81a611..2f84054c9b7d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -74,7 +74,6 @@ struct adau1373 { #define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) #define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) #define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) -#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) #define ADAU1373_HEADDECT 0x36 #define ADAU1373_ADC_DAC_STATUS 0x37 #define ADAU1373_ADC_CTRL 0x3c -- cgit From 729485f6adbf1c7e1e08a01d2c276da30a91b0a4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 13:47:10 +0200 Subject: ASoC: adau1373: Issue soft reset on probe Reset the device on probe to make sure that the register settings match the register cache defaults. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 2f84054c9b7d..59654b1e7f3f 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1540,6 +1540,8 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (IS_ERR(adau1373->regmap)) return PTR_ERR(adau1373->regmap); + regmap_write(adau1373->regmap, ADAU1373_SOFT_RESET, 0x00); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, -- cgit From 2560b3d1bdf1344aa65bba1523a08e4db27a3c14 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 27 Sep 2013 15:18:25 +0200 Subject: ASoC: adav80x: Convert to direct regmap usage Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 147 +++++++++++++++++++++++++++++++-------------- 1 file changed, 102 insertions(+), 45 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0f226..14a7c169d004 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,34 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static struct reg_default adav80x_reg_defaults[] = { + { ADAV80X_PLAYBACK_CTRL, 0x01 }, + { ADAV80X_AUX_IN_CTRL, 0x01 }, + { ADAV80X_REC_CTRL, 0x02 }, + { ADAV80X_AUX_OUT_CTRL, 0x01 }, + { ADAV80X_DPATH_CTRL1, 0xc0 }, + { ADAV80X_DPATH_CTRL2, 0x11 }, + { ADAV80X_DAC_CTRL1, 0x00 }, + { ADAV80X_DAC_CTRL2, 0x00 }, + { ADAV80X_DAC_CTRL3, 0x00 }, + { ADAV80X_DAC_L_VOL, 0xff }, + { ADAV80X_DAC_R_VOL, 0xff }, + { ADAV80X_PGA_L_VOL, 0x00 }, + { ADAV80X_PGA_R_VOL, 0x00 }, + { ADAV80X_ADC_CTRL1, 0x00 }, + { ADAV80X_ADC_CTRL2, 0x00 }, + { ADAV80X_ADC_L_VOL, 0xff }, + { ADAV80X_ADC_R_VOL, 0xff }, + { ADAV80X_PLL_CTRL1, 0x00 }, + { ADAV80X_PLL_CTRL2, 0x00 }, + { ADAV80X_ICLK_CTRL1, 0x00 }, + { ADAV80X_ICLK_CTRL2, 0x00 }, + { ADAV80X_PLL_CLK_SRC, 0x00 }, + { ADAV80X_PLL_OUTE, 0x00 }, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -298,7 +310,7 @@ static int adav80x_set_deemph(struct snd_soc_codec *codec) val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; } - return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + return regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); } @@ -394,10 +406,11 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, capture); - snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + regmap_write(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], + playback); adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -407,6 +420,7 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int adav80x_set_adc_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -414,7 +428,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, else val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; - snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + regmap_update_bits(adav80x->regmap, ADAV80X_ADC_CTRL1, ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); return 0; @@ -423,6 +437,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, static int adav80x_set_dac_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -430,7 +445,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, else val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, val); @@ -440,6 +455,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, struct snd_soc_dai *dai, snd_pcm_format_t format) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; switch (format) { @@ -459,7 +475,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_WORD_LEN_MASK, val); return 0; @@ -491,7 +507,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], ADAV80X_PLAYBACK_MODE_MASK, val); return 0; @@ -554,8 +570,10 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); - snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); - snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL1, + iclk_ctrl1); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, + iclk_ctrl2); snd_soc_dapm_sync(&codec->dapm); } @@ -575,10 +593,12 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); if (freq == 0) { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, mask); adav80x->sysclk_pd[clk_id] = true; } else { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, 0); adav80x->sysclk_pd[clk_id] = false; } @@ -650,9 +670,9 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, - pll_ctrl1); - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL1, + ADAV80X_PLL_CTRL1_PLLDIV, pll_ctrl1); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL2, ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); if (source != adav80x->pll_src) { @@ -661,7 +681,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, else pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); - snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CLK_SRC, ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); adav80x->pll_src = source; @@ -675,6 +695,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, static int adav80x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int mask = ADAV80X_DAC_CTRL1_PD; switch (level) { @@ -683,10 +704,12 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + 0x00); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + mask); break; } @@ -780,7 +803,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -791,23 +814,31 @@ static int adav80x_probe(struct snd_soc_codec *codec) snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); /* Power down S/PDIF receiver, since it is currently not supported */ - snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20); /* Disable DAC zero flag */ - snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +858,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,18 +866,21 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) +static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; + dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; + adav80x->regmap = regmap; ret = snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); @@ -868,6 +898,19 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct regmap_config adav80x_spi_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + .read_flag_mask = 0x01, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, { } @@ -876,7 +919,8 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + return adav80x_bus_probe(&spi->dev, + devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); } static int adav80x_spi_remove(struct spi_device *spi) @@ -896,6 +940,18 @@ static struct spi_driver adav80x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct regmap_config adav80x_i2c_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } @@ -905,7 +961,8 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); } static int adav80x_i2c_remove(struct i2c_client *client) -- cgit From 648c538204c23370c734d72921155cc24aff928d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:48:24 +0200 Subject: ASoC: tas5086: move two variables into private struct We need to access the charge_period and start_mid_z values from other places later, so move them to the private struct. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88f7204..31b5868ef7c1 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -244,6 +244,8 @@ struct tas5086_private { unsigned int mclk, sclk; unsigned int format; bool deemph; + unsigned int charge_period; + unsigned int pwm_start_mid_z; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -720,13 +722,15 @@ static const int tas5086_charge_period[] = { static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start_mid_z = 0; int i, ret; + priv->pwm_start_mid_z = 0; + priv->charge_period = 1300000; /* hardware default is 1300 ms */ + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; - of_property_read_u32(of_node, "ti,charge-period", &charge_period); + of_property_read_u32(of_node, "ti,charge-period", + &priv->charge_period); for (i = 0; i < 6; i++) { char name[25]; @@ -735,7 +739,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start_mid_z |= 1 << i; + priv->pwm_start_mid_z |= 1 << i; } } @@ -744,25 +748,25 @@ static int tas5086_probe(struct snd_soc_codec *codec) * configure 'part 1' of the PWM starts to use Mid-Z, and tell * all configured mid-z channels to start start under 'part 1'. */ - if (pwm_start_mid_z) + if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, TAS5086_PWM_START_MIDZ_FOR_START_1 | - pwm_start_mid_z); + priv->pwm_start_mid_z); /* lookup and set split-capacitor charge period */ - if (charge_period == 0) { + if (priv->charge_period == 0) { regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); } else { i = index_in_array(tas5086_charge_period, ARRAY_SIZE(tas5086_charge_period), - charge_period); + priv->charge_period); if (i >= 0) regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, i + 0x08); else dev_warn(codec->dev, "Invalid split-cap charge period of %d ns.\n", - charge_period); + priv->charge_period); } /* enable factory trim */ -- cgit From d5fd3ccc2d9df493ad6f1eaf7aba72f690e98937 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:48:25 +0200 Subject: ASoC: tas5086: move initialization code to own functions We'll need to call code to initialize and reset the codec again at resume time, so factor it out first. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 128 +++++++++++++++++++++++++-------------------- 1 file changed, 72 insertions(+), 56 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 31b5868ef7c1..3a88c68145c2 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -458,6 +458,75 @@ static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream) return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val); } +static void tas5086_reset(struct tas5086_private *priv) +{ + if (gpio_is_valid(priv->gpio_nreset)) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(priv->gpio_nreset, 0); + udelay(1); + gpio_set_value(priv->gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +} + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_init(struct device *dev, struct tas5086_private *priv) +{ + int ret, i; + + /* + * If any of the channels is configured to start in Mid-Z mode, + * configure 'part 1' of the PWM starts to use Mid-Z, and tell + * all configured mid-z channels to start start under 'part 1'. + */ + if (priv->pwm_start_mid_z) + regmap_write(priv->regmap, TAS5086_PWM_START, + TAS5086_PWM_START_MIDZ_FOR_START_1 | + priv->pwm_start_mid_z); + + /* lookup and set split-capacitor charge period */ + if (priv->charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + priv->charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(dev, + "Invalid split-cap charge period of %d ns.\n", + priv->charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* mute all channels for now */ + ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + TAS5086_SOFT_MUTE_ALL); + if (ret < 0) + return ret; + + return 0; +} + /* TAS5086 controls */ static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); @@ -712,13 +781,6 @@ static const struct of_device_id tas5086_dt_ids[] = { MODULE_DEVICE_TABLE(of, tas5086_dt_ids); #endif -/* charge period values in microseconds */ -static const int tas5086_charge_period[] = { - 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, - 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, - 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, -}; - static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); @@ -729,6 +791,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; + of_property_read_u32(of_node, "ti,charge-period", &priv->charge_period); @@ -743,39 +806,7 @@ static int tas5086_probe(struct snd_soc_codec *codec) } } - /* - * If any of the channels is configured to start in Mid-Z mode, - * configure 'part 1' of the PWM starts to use Mid-Z, and tell - * all configured mid-z channels to start start under 'part 1'. - */ - if (priv->pwm_start_mid_z) - regmap_write(priv->regmap, TAS5086_PWM_START, - TAS5086_PWM_START_MIDZ_FOR_START_1 | - priv->pwm_start_mid_z); - - /* lookup and set split-capacitor charge period */ - if (priv->charge_period == 0) { - regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); - } else { - i = index_in_array(tas5086_charge_period, - ARRAY_SIZE(tas5086_charge_period), - priv->charge_period); - if (i >= 0) - regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, - i + 0x08); - else - dev_warn(codec->dev, - "Invalid split-cap charge period of %d ns.\n", - priv->charge_period); - } - - /* enable factory trim */ - ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); - if (ret < 0) - return ret; - - /* start all channels */ - ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + ret = tas5086_init(codec->dev, priv); if (ret < 0) return ret; @@ -784,12 +815,6 @@ static int tas5086_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - /* mute all channels for now */ - ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, - TAS5086_SOFT_MUTE_ALL); - if (ret < 0) - return ret; - return 0; } @@ -866,17 +891,8 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset")) gpio_nreset = -EINVAL; - if (gpio_is_valid(gpio_nreset)) { - /* Reset codec - minimum assertion time is 400ns */ - gpio_direction_output(gpio_nreset, 0); - udelay(1); - gpio_set_value(gpio_nreset, 1); - - /* Codec needs ~15ms to wake up */ - msleep(15); - } - priv->gpio_nreset = gpio_nreset; + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); -- cgit From 25c84cc1ace56421fa9a676a387a1919e7bc4e62 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 1 Oct 2013 14:48:26 +0200 Subject: ASoC: tas5086: add suspend callback When going to suspend, shut down all channels and re-do the init procedure at resume time. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 3a88c68145c2..2996d2ea026b 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -762,14 +762,39 @@ static struct snd_soc_dai_driver tas5086_dai = { }; #ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + /* Shut down all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x60); + if (ret < 0) + return ret; + + return 0; +} + static int tas5086_soc_resume(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; - /* Restore codec state */ - return regcache_sync(priv->regmap); + tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + + ret = tas5086_init(codec->dev, priv); + if (ret < 0) + return ret; + + ret = regcache_sync(priv->regmap); + if (ret < 0) + return ret; + + return 0; } #else +#define tas5086_soc_suspend NULL #define tas5086_soc_resume NULL #endif /* CONFIG_PM */ @@ -832,6 +857,7 @@ static int tas5086_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { .probe = tas5086_probe, .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, .resume = tas5086_soc_resume, .controls = tas5086_controls, .num_controls = ARRAY_SIZE(tas5086_controls), -- cgit From 290c348ee5522a5682c2011fa4d51f232404e8a4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Oct 2013 13:43:51 +0200 Subject: ASoC: twl6040: Use virtual DAPM mixer controls By using the new virtual DAPM mixer controls it is possible to remove the twl6040 specific implementation of virtual controls. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 35059a242fa4..f2f4bcb2ff71 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -54,12 +54,7 @@ enum twl6040_dai_id { #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -/* Shadow register used by the driver */ -#define TWL6040_REG_SW_SHADOW 0x2F -#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) - -/* TWL6040_REG_SW_SHADOW (0x2F) fields */ -#define TWL6040_EAR_PATH_ENABLE 0x01 +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) struct twl6040_jack_data { struct snd_soc_jack *jack; @@ -135,8 +130,6 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* REG_HFOTRIM 0x2C */ 0x09, /* REG_ACCCTL 0x2D */ 0x00, /* REG_STATUS 0x2E (ro) */ - - 0x00, /* REG_SW_SHADOW 0x2F - Shadow, non HW register */ }; /* List of registers to be restored after power up */ @@ -220,12 +213,8 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - if (likely(reg < TWL6040_REG_SW_SHADOW)) { - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); - } else { - value = twl6040_read_reg_cache(codec, reg); - } + value = twl6040_reg_read(twl6040, reg); + twl6040_write_reg_cache(codec, reg, value); return value; } @@ -261,8 +250,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (likely(reg < TWL6040_REG_SW_SHADOW) && - twl6040_is_path_unmuted(codec, reg)) + if (twl6040_is_path_unmuted(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; @@ -555,7 +543,7 @@ static const struct snd_kcontrol_new hfr_mux_controls = SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]); static const struct snd_kcontrol_new ep_path_enable_control = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); static const struct snd_kcontrol_new auxl_switch_control = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0); -- cgit From 052901f42f360062f36cc5c0aa6e5ae372fe0895 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Oct 2013 13:43:50 +0200 Subject: ASoC: twl4030: Use virtual DAPM mixer controls By using the new virtual DAPM mixer controls it is possible to remove the twl4030 specific implementation of virtual controls. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 80 +++++++++++++++++++++------------------------- 1 file changed, 36 insertions(+), 44 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1e3884d6b3fb..dfc51bb425da 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,13 +46,7 @@ /* TWL4030 PMBR1 Register GPIO6 mux bits */ #define TWL4030_GPIO6_PWM0_MUTE(value) ((value & 0x03) << 2) -/* Shadow register used by the audio driver */ -#define TWL4030_REG_SW_SHADOW 0x4A -#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) - -/* TWL4030_REG_SW_SHADOW (0x4A) Fields */ -#define TWL4030_HFL_EN 0x01 -#define TWL4030_HFR_EN 0x02 +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) /* * twl4030 register cache & default register settings @@ -132,7 +126,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_PWM_SET (0x47) */ 0x00, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ - 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; /* codec private data */ @@ -198,42 +191,41 @@ static int twl4030_write(struct snd_soc_codec *codec, int write_to_reg = 0; twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) { - /* Decide if the given register can be written */ - switch (reg) { - case TWL4030_REG_EAR_CTL: - if (twl4030->earpiece_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PREDL_CTL: - if (twl4030->predrivel_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PREDR_CTL: - if (twl4030->predriver_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PRECKL_CTL: - if (twl4030->carkitl_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PRECKR_CTL: - if (twl4030->carkitr_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_HS_GAIN_SET: - if (twl4030->hsl_enabled || twl4030->hsr_enabled) - write_to_reg = 1; - break; - default: - /* All other register can be written */ + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) write_to_reg = 1; - break; - } - if (write_to_reg) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - value, reg); + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + return 0; } @@ -532,7 +524,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Left virtual mute */ static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = - SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = @@ -548,7 +540,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); /* Handsfree Right virtual mute */ static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = - SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); /* Vibra */ /* Vibra audio path selection */ -- cgit From c6452e39e8286b88872aee20a4d083cfa65516bc Mon Sep 17 00:00:00 2001 From: Steffen Trumtrar Date: Fri, 11 Oct 2013 12:28:13 +0200 Subject: ASoC: mc13783: add mixer controls Add more controls to the alsa mixer infrastructure. Signed-off-by: Steffen Trumtrar Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index eedbf05b8e96..2b62737bf3d4 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -541,8 +541,26 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) -- cgit From bb7838d4f13c50df8ef7324f5fd4aeb729269e22 Mon Sep 17 00:00:00 2001 From: Steffen Trumtrar Date: Fri, 11 Oct 2013 12:28:14 +0200 Subject: ASoC: mc13783: add more DAPM routes Add more infrastructure (i.e. routes, muxes, switches) to the mc13783 DAPM. Signed-off-by: Steffen Trumtrar Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 52 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 2b62737bf3d4..f5472adee674 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -427,6 +427,29 @@ static const struct snd_kcontrol_new right_input_mux = static const struct snd_kcontrol_new samp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); +static const char * const speaker_amp_source_text[] = { + "CODEC", "Right" +}; +static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, + speaker_amp_source_text); +static const struct snd_kcontrol_new speaker_amp_source_mux = + SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); + +static const char * const headset_amp_source_text[] = { + "CODEC", "Mixer" +}; + +static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, + headset_amp_source_text); +static const struct snd_kcontrol_new headset_amp_source_mux = + SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); + +static const struct snd_kcontrol_new cdcout_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0); + +static const struct snd_kcontrol_new adc_bypass_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_CODEC, 16, 1, 0); + static const struct snd_kcontrol_new lamp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); @@ -464,12 +487,22 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, &right_input_mux), + SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0, + &speaker_amp_source_mux), + + SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0, + &headset_amp_source_mux), + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), + SND_SOC_DAPM_PGA("Voice CODEC PGA", MC13783_AUDIO_RX1, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Voice CODEC Bypass", MC13783_AUDIO_CODEC, 16, 0, + &adc_bypass_ctl), + /* Output */ SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), @@ -477,10 +510,15 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSPL"), SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), + SND_SOC_DAPM_OUTPUT("CDCOUT"), - SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, + &cdcout_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, &hlamp_ctl), @@ -515,20 +553,28 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "ADC", NULL, "PGA Right Input"}, { "ADC", NULL, "ADC_Reset"}, + { "Voice CODEC PGA", "Voice CODEC Bypass", "ADC" }, + + { "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Speaker Amp Source MUX", "Right", "DAC PGA"}, + + { "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Headset Amp Source MUX", "Mixer", "DAC PGA"}, + /* Output */ { "HSL", NULL, "Headset Amp Left" }, { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", NULL, "Speaker Amp"}, - { "Speaker Amp", NULL, "DAC PGA"}, - { "LSP", NULL, "DAC PGA"}, - { "Headset Amp Left", NULL, "DAC PGA"}, - { "Headset Amp Right", NULL, "DAC PGA"}, + { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"}, + { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, + { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, { "Line out Amp Left", NULL, "DAC PGA"}, { "Line out Amp Right", NULL, "DAC PGA"}, { "DAC PGA", NULL, "DAC"}, { "DAC", NULL, "DAC_E"}, + { "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"}, }; static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", -- cgit From a7ea1b7249adc8c090a0b277ab5f3737ee4023c1 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:56 +0530 Subject: ASoC: cs4271: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb8f071..f6e953454bc0 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include -- cgit From 193a47162c93afa09fffd04a04443f14d402c606 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:57 +0530 Subject: ASoC: pcm1681: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 651ce0923675..54ea15b87bfc 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include -- cgit From 4b2fa5121c758db6fe9ed4931b54e390661395de Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:58 +0530 Subject: ASoC: pcm1792a: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 2a8eccf64c76..6f14c50a7f0f 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include "pcm1792a.h" -- cgit From 285d00c11b0a8d0ef63c176f88caab5071c9e80d Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:23:59 +0530 Subject: ASoC: tas5086: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 2996d2ea026b..fe4d29d88564 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -37,6 +37,7 @@ #include #include #include +#include #include #include #include -- cgit From b3b70786ec18ef3088b55b76258bbd48d75aee08 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Fri, 11 Oct 2013 17:24:00 +0530 Subject: ASoC: tlv320aic3x: Include linux/of.h header 'of_match_ptr' is defined in linux/of.h. Include it explicitly. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 892c108ca67a..f8b9fa6b6f0a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include -- cgit From 3d8c8bc0250f7cb11f887691b7473b51adcd2bcb Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Oct 2013 11:03:33 -0500 Subject: ASoC: cs42l73: Add platform data support for cs42l73 codec Add support for RST GPIO and Charge Pump Freq in platform data Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 51 ++++++++++++++++++++++++++++++---------------- sound/soc/codecs/cs42l73.h | 1 + 2 files changed, 34 insertions(+), 18 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3b20c86cdb01..db9d39604d68 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -28,6 +29,7 @@ #include #include #include +#include #include "cs42l73.h" struct sp_config { @@ -35,6 +37,7 @@ struct sp_config { u32 srate; }; struct cs42l73_private { + struct cs42l73_platform_data pdata; struct sp_config config[3]; struct regmap *regmap; u32 sysclk; @@ -310,15 +313,6 @@ static const struct soc_enum ng_delay_enum = SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); -static const char * const charge_pump_freq_text[] = { - "0", "1", "2", "3", "4", - "5", "6", "7", "8", "9", - "10", "11", "12", "13", "14", "15" }; - -static const struct soc_enum charge_pump_enum = - SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, - ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); - static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -511,8 +505,6 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), SOC_ENUM("NG Delay", ng_delay_enum), - SOC_ENUM("Charge Pump Frequency", charge_pump_enum), - SOC_DOUBLE_R_TLV("XSP-IP Volume", CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, attn_tlv), @@ -1367,11 +1359,16 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(cs42l73->regmap, true); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + /* Set Charge Pump Frequency */ + if (cs42l73->pdata.chgfreq) + snd_soc_update_bits(codec, CS42L73_CPFCHC, + CS42L73_CHARGEPUMP_MASK, + cs42l73->pdata.chgfreq << 4); + + /* MCLK1 as master clk */ + cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; return ret; @@ -1415,6 +1412,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l73_private *cs42l73; + struct cs42l73_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1426,14 +1424,32 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } - i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } + + if (pdata) + cs42l73->pdata = *pdata; + + i2c_set_clientdata(i2c_client, cs42l73); + + if (cs42l73->pdata.reset_gpio) { + ret = gpio_request_one(cs42l73->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L73 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l73->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 1); + } + + regcache_cache_bypass(cs42l73->regmap, true); + /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); devid = (reg & 0xFF) << 12; @@ -1444,7 +1460,6 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, ®); devid |= (reg & 0xF0) >> 4; - if (devid != CS42L73_DEVID) { ret = -ENODEV; dev_err(&i2c_client->dev, @@ -1462,7 +1477,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "Cirrus Logic CS42L73, Revision: %02X\n", reg & 0xFF); - regcache_cache_only(cs42l73->regmap, true); + regcache_cache_bypass(cs42l73->regmap, false); ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l73, cs42l73_dai, diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index f30a4c4d62e6..4f83d39496a8 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -159,6 +159,7 @@ #define THMOVLD_115C 2 #define THMOVLD_098C 3 +#define CS42L73_CHARGEPUMP_MASK (0xF0) /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ #define SP_3ST (1 << 7) -- cgit From f9ca060680e7c26a88d990ad9370572274b0d54b Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Thu, 17 Oct 2013 11:03:34 -0500 Subject: ASoC: cs42l73: Namespace defines for cs42l73 codec Cleanup to namespace the defines for the cs42l73 driver Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 38 ++++++++--------- sound/soc/codecs/cs42l73.h | 104 ++++++++++++++++++++++----------------------- 2 files changed, 70 insertions(+), 72 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index db9d39604d68..89efc3c6aefc 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1047,11 +1047,11 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - mmcc |= MS_MASTER; + mmcc |= CS42L73_MS_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - mmcc &= ~MS_MASTER; + mmcc &= ~CS42L73_MS_MASTER; break; default: @@ -1063,11 +1063,11 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (format) { case SND_SOC_DAIFMT_I2S: - spc &= ~SPDIF_PCM; + spc &= ~CS42L73_SPDIF_PCM; break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - if (mmcc & MS_MASTER) { + if (mmcc & CS42L73_MS_MASTER) { dev_err(codec->dev, "PCM format in slave mode only\n"); return -EINVAL; @@ -1077,25 +1077,25 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) "PCM format is not supported on ASP port\n"); return -EINVAL; } - spc |= SPDIF_PCM; + spc |= CS42L73_SPDIF_PCM; break; default: return -EINVAL; } - if (spc & SPDIF_PCM) { + if (spc & CS42L73_SPDIF_PCM) { /* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */ - spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER); + spc &= ~(CS42L73_PCM_MODE_MASK | CS42L73_PCM_BIT_ORDER); switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= PCM_MODE0; + spc |= CS42L73_PCM_MODE0; if (inv == SND_SOC_DAIFMT_IB_NF) - spc |= PCM_MODE1; + spc |= CS42L73_PCM_MODE1; break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= PCM_MODE1; + spc |= CS42L73_PCM_MODE1; break; default: return -EINVAL; @@ -1155,7 +1155,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, int mclk_coeff; int srate = params_rate(params); - if (priv->config[id].mmcc & MS_MASTER) { + if (priv->config[id].mmcc & CS42L73_MS_MASTER) { /* CS42L73 Master */ /* MCLK -> srate */ mclk_coeff = @@ -1174,13 +1174,13 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].spc &= 0xFC; /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ if (priv->mclk >= 6400000) - priv->config[id].spc |= MCK_SCLK_64FS; + priv->config[id].spc |= CS42L73_MCK_SCLK_64FS; else - priv->config[id].spc |= MCK_SCLK_MCLK; + priv->config[id].spc |= CS42L73_MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_64FS; + priv->config[id].spc |= CS42L73_MCK_SCLK_64FS; } /* Update ASRCs */ priv->config[id].srate = srate; @@ -1200,8 +1200,8 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 0); break; case SND_SOC_BIAS_PREPARE: @@ -1212,11 +1212,11 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(cs42l73->regmap, false); regcache_sync(cs42l73->regmap); } - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1); if (cs42l73->shutdwn_delay > 0) { mdelay(cs42l73->shutdwn_delay); cs42l73->shutdwn_delay = 0; @@ -1225,7 +1225,7 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, * down. */ } - snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 1); break; } codec->dapm.bias_level = level; diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index 4f83d39496a8..45746186a678 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -128,60 +128,60 @@ /* Bitfield Definitions */ /* CS42L73_PWRCTL1 */ -#define PDN_ADCB (1 << 7) -#define PDN_DMICB (1 << 6) -#define PDN_ADCA (1 << 5) -#define PDN_DMICA (1 << 4) -#define PDN_LDO (1 << 2) -#define DISCHG_FILT (1 << 1) -#define PDN (1 << 0) +#define CS42L73_PDN_ADCB (1 << 7) +#define CS42L73_PDN_DMICB (1 << 6) +#define CS42L73_PDN_ADCA (1 << 5) +#define CS42L73_PDN_DMICA (1 << 4) +#define CS42L73_PDN_LDO (1 << 2) +#define CS42L73_DISCHG_FILT (1 << 1) +#define CS42L73_PDN (1 << 0) /* CS42L73_PWRCTL2 */ -#define PDN_MIC2_BIAS (1 << 7) -#define PDN_MIC1_BIAS (1 << 6) -#define PDN_VSP (1 << 4) -#define PDN_ASP_SDOUT (1 << 3) -#define PDN_ASP_SDIN (1 << 2) -#define PDN_XSP_SDOUT (1 << 1) -#define PDN_XSP_SDIN (1 << 0) +#define CS42L73_PDN_MIC2_BIAS (1 << 7) +#define CS42L73_PDN_MIC1_BIAS (1 << 6) +#define CS42L73_PDN_VSP (1 << 4) +#define CS42L73_PDN_ASP_SDOUT (1 << 3) +#define CS42L73_PDN_ASP_SDIN (1 << 2) +#define CS42L73_PDN_XSP_SDOUT (1 << 1) +#define CS42L73_PDN_XSP_SDIN (1 << 0) /* CS42L73_PWRCTL3 */ -#define PDN_THMS (1 << 5) -#define PDN_SPKLO (1 << 4) -#define PDN_EAR (1 << 3) -#define PDN_SPK (1 << 2) -#define PDN_LO (1 << 1) -#define PDN_HP (1 << 0) +#define CS42L73_PDN_THMS (1 << 5) +#define CS42L73_PDN_SPKLO (1 << 4) +#define CS42L73_PDN_EAR (1 << 3) +#define CS42L73_PDN_SPK (1 << 2) +#define CS42L73_PDN_LO (1 << 1) +#define CS42L73_PDN_HP (1 << 0) /* Thermal Overload Detect. Requires interrupt ... */ -#define THMOVLD_150C 0 -#define THMOVLD_132C 1 -#define THMOVLD_115C 2 -#define THMOVLD_098C 3 +#define CS42L73_THMOVLD_150C 0 +#define CS42L73_THMOVLD_132C 1 +#define CS42L73_THMOVLD_115C 2 +#define CS42L73_THMOVLD_098C 3 #define CS42L73_CHARGEPUMP_MASK (0xF0) /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ -#define SP_3ST (1 << 7) -#define SPDIF_I2S (0 << 6) -#define SPDIF_PCM (1 << 6) -#define PCM_MODE0 (0 << 4) -#define PCM_MODE1 (1 << 4) -#define PCM_MODE2 (2 << 4) -#define PCM_MODE_MASK (3 << 4) -#define PCM_BIT_ORDER (1 << 3) -#define MCK_SCLK_64FS (0 << 0) -#define MCK_SCLK_MCLK (2 << 0) -#define MCK_SCLK_PREMCLK (3 << 0) +#define CS42L73_SP_3ST (1 << 7) +#define CS42L73_SPDIF_I2S (0 << 6) +#define CS42L73_SPDIF_PCM (1 << 6) +#define CS42L73_PCM_MODE0 (0 << 4) +#define CS42L73_PCM_MODE1 (1 << 4) +#define CS42L73_PCM_MODE2 (2 << 4) +#define CS42L73_PCM_MODE_MASK (3 << 4) +#define CS42L73_PCM_BIT_ORDER (1 << 3) +#define CS42L73_MCK_SCLK_64FS (0 << 0) +#define CS42L73_MCK_SCLK_MCLK (2 << 0) +#define CS42L73_MCK_SCLK_PREMCLK (3 << 0) /* CS42L73_xSPMMCC */ -#define MS_MASTER (1 << 7) +#define CS42L73_MS_MASTER (1 << 7) /* CS42L73_DMMCC */ -#define MCLKDIS (1 << 0) -#define MCLKSEL_MCLK2 (1 << 4) -#define MCLKSEL_MCLK1 (0 << 4) +#define CS42L73_MCLKDIS (1 << 0) +#define CS42L73_MCLKSEL_MCLK2 (1 << 4) +#define CS42L73_MCLKSEL_MCLK1 (0 << 4) /* CS42L73 MCLK derived from MCLK1 or MCLK2 */ #define CS42L73_CLKID_MCLK1 0 @@ -195,28 +195,26 @@ #define CS42L73_VSP 2 /* IS1, IM1 */ -#define MIC2_SDET (1 << 6) -#define THMOVLD (1 << 4) -#define DIGMIXOVFL (1 << 3) -#define IPBOVFL (1 << 1) -#define IPAOVFL (1 << 0) +#define CS42L73_MIC2_SDET (1 << 6) +#define CS42L73_THMOVLD (1 << 4) +#define CS42L73_DIGMIXOVFL (1 << 3) +#define CS42L73_IPBOVFL (1 << 1) +#define CS42L73_IPAOVFL (1 << 0) /* Analog Softramp */ -#define ANLGOSFT (1 << 0) +#define CS42L73_ANLGOSFT (1 << 0) /* HP A/B Analog Mute */ -#define HPA_MUTE (1 << 7) +#define CS42L73_HPA_MUTE (1 << 7) /* LO A/B Analog Mute */ -#define LOA_MUTE (1 << 7) +#define CS42L73_LOA_MUTE (1 << 7) /* Digital Mute */ -#define HLAD_MUTE (1 << 0) -#define HLBD_MUTE (1 << 1) -#define SPKD_MUTE (1 << 2) -#define ESLD_MUTE (1 << 3) +#define CS42L73_HLAD_MUTE (1 << 0) +#define CS42L73_HLBD_MUTE (1 << 1) +#define CS42L73_SPKD_MUTE (1 << 2) +#define CS42L73_ESLD_MUTE (1 << 3) /* Misc defines for codec */ -#define CS42L73_RESET_GPIO 143 - #define CS42L73_DEVID 0x00042A73 #define CS42L73_MCLKX_MIN 5644800 #define CS42L73_MCLKX_MAX 38400000 -- cgit From ca50410b731c636b9750c02d5ae45be215056634 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 14:56:13 +0100 Subject: ASoC: wm8962: Move interrupt initalisation to probe() This is more idiomatic and fixes bugs in the error handling paths. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 68 +++++++++++++++++++++++------------------------ 1 file changed, 33 insertions(+), 35 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11d80f3b6137..54379ee1cd0c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = &wm8962->pdata; - int i, trigger, irq_pol; + int i; bool dmicclk, dmicdat; wm8962->codec = codec; @@ -3506,36 +3506,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_init_beep(codec); wm8962_init_gpio(codec); - if (wm8962->irq) { - if (pdata->irq_active_low) { - trigger = IRQF_TRIGGER_LOW; - irq_pol = WM8962_IRQ_POL; - } else { - trigger = IRQF_TRIGGER_HIGH; - irq_pol = 0; - } - - snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL, - WM8962_IRQ_POL, irq_pol); - - ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq, - trigger | IRQF_ONESHOT, - "wm8962", codec->dev); - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ %d: %d\n", - wm8962->irq, ret); - wm8962->irq = 0; - /* Non-fatal */ - } else { - /* Enable some IRQs by default */ - snd_soc_update_bits(codec, - WM8962_INTERRUPT_STATUS_2_MASK, - WM8962_FLL_LOCK_EINT | - WM8962_TEMP_SHUT_EINT | - WM8962_FIFOS_ERR_EINT, 0); - } - } - return 0; } @@ -3544,9 +3514,6 @@ static int wm8962_remove(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int i; - if (wm8962->irq) - free_irq(wm8962->irq, codec); - cancel_delayed_work_sync(&wm8962->mic_work); wm8962_free_gpio(codec); @@ -3619,7 +3586,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, struct wm8962_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm8962_priv *wm8962; unsigned int reg; - int ret, i; + int ret, i, irq_pol, trigger; wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), GFP_KERNEL); @@ -3714,6 +3681,37 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ret); } + if (wm8962->irq) { + if (pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8962_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + regmap_update_bits(wm8962->regmap, WM8962_INTERRUPT_CONTROL, + WM8962_IRQ_POL, irq_pol); + + ret = devm_request_threaded_irq(&i2c->dev, wm8962->irq, NULL, + wm8962_irq, + trigger | IRQF_ONESHOT, + "wm8962", &i2c->dev); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n", + wm8962->irq, ret); + wm8962->irq = 0; + /* Non-fatal */ + } else { + /* Enable some IRQs by default */ + regmap_update_bits(wm8962->regmap, + WM8962_INTERRUPT_STATUS_2_MASK, + WM8962_FLL_LOCK_EINT | + WM8962_TEMP_SHUT_EINT | + WM8962_FIFOS_ERR_EINT, 0); + } + } + pm_runtime_enable(&i2c->dev); pm_request_idle(&i2c->dev); -- cgit From 78b78f5c019e5c68c88afad4b0d3070becde939e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Oct 2013 15:04:21 +0100 Subject: ASoC: wm8962: Move register initialisation to I2C probe() This is more idiomatic and is required for robust operation since we must ensure that the clocking configuration is valid as rapidly as possible. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 150 +++++++++++++++++++++++----------------------- 1 file changed, 75 insertions(+), 75 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 54379ee1cd0c..2bf9ee7c5407 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3242,7 +3242,7 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) } #endif -static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio) +static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) { int mask = 0; int val = 0; @@ -3263,8 +3263,8 @@ static void wm8962_set_gpio_mode(struct snd_soc_codec *codec, int gpio) } if (mask) - snd_soc_update_bits(codec, WM8962_ANALOGUE_CLOCKING1, - mask, val); + regmap_update_bits(wm8962->regmap, WM8962_ANALOGUE_CLOCKING1, + mask, val); } #ifdef CONFIG_GPIOLIB @@ -3276,7 +3276,6 @@ static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) { struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); - struct snd_soc_codec *codec = wm8962->codec; /* The WM8962 GPIOs aren't linearly numbered. For simplicity * we export linear numbers and error out if the unsupported @@ -3292,7 +3291,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) return -EINVAL; } - wm8962_set_gpio_mode(codec, offset + 1); + wm8962_set_gpio_mode(wm8962, offset + 1); return 0; } @@ -3376,7 +3375,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = &wm8962->pdata; int i; bool dmicclk, dmicdat; @@ -3409,75 +3407,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) } } - /* SYSCLK defaults to on; make sure it is off so we can safely - * write to registers if the device is declocked. - */ - snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); - - /* Ensure we have soft control over all registers */ - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); - - /* Ensure that the oscillator and PLLs are disabled */ - snd_soc_update_bits(codec, WM8962_PLL2, - WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, - 0); - - /* Apply static configuration for GPIOs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) - if (pdata->gpio_init[i]) { - wm8962_set_gpio_mode(codec, i + 1); - snd_soc_write(codec, 0x200 + i, - pdata->gpio_init[i] & 0xffff); - } - - - /* Put the speakers into mono mode? */ - if (pdata->spk_mono) - snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2, - WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); - - /* Micbias setup, detection enable and detection - * threasholds. */ - if (pdata->mic_cfg) - snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_4, - WM8962_MICDET_ENA | - WM8962_MICDET_THR_MASK | - WM8962_MICSHORT_THR_MASK | - WM8962_MICBIAS_LVL, - pdata->mic_cfg); - - /* Latch volume update bits */ - snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, - WM8962_IN_VU, WM8962_IN_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, - WM8962_IN_VU, WM8962_IN_VU); - snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, - WM8962_ADC_VU, WM8962_ADC_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, - WM8962_ADC_VU, WM8962_ADC_VU); - snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, - WM8962_DAC_VU, WM8962_DAC_VU); - snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, - WM8962_DAC_VU, WM8962_DAC_VU); - snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, - WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); - snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, - WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); - snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, - WM8962_HPOUT_VU, WM8962_HPOUT_VU); - snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, - WM8962_HPOUT_VU, WM8962_HPOUT_VU); - - /* Stereo control for EQ */ - snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0); - - /* Don't debouce interrupts so we don't need SYSCLK */ - snd_soc_update_bits(codec, WM8962_IRQ_DEBOUNCE, - WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | - WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, - 0); - wm8962_add_widgets(codec); /* Save boards having to disable DMIC when not in use */ @@ -3671,6 +3600,77 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, goto err_enable; } + /* SYSCLK defaults to on; make sure it is off so we can safely + * write to registers if the device is declocked. + */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA, 0); + + /* Ensure we have soft control over all registers */ + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + + /* Ensure that the oscillator and PLLs are disabled */ + regmap_update_bits(wm8962->regmap, WM8962_PLL2, + WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA, + 0); + + /* Apply static configuration for GPIOs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_init); i++) + if (pdata->gpio_init[i]) { + wm8962_set_gpio_mode(wm8962, i + 1); + regmap_write(wm8962->regmap, 0x200 + i, + pdata->gpio_init[i] & 0xffff); + } + + + /* Put the speakers into mono mode? */ + if (pdata->spk_mono) + regmap_update_bits(wm8962->regmap, WM8962_CLASS_D_CONTROL_2, + WM8962_SPK_MONO_MASK, WM8962_SPK_MONO); + + /* Micbias setup, detection enable and detection + * threasholds. */ + if (pdata->mic_cfg) + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_MICDET_ENA | + WM8962_MICDET_THR_MASK | + WM8962_MICSHORT_THR_MASK | + WM8962_MICBIAS_LVL, + pdata->mic_cfg); + + /* Latch volume update bits */ + regmap_update_bits(wm8962->regmap, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + regmap_update_bits(wm8962->regmap, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + regmap_update_bits(wm8962->regmap, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + regmap_update_bits(wm8962->regmap, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + regmap_update_bits(wm8962->regmap, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + regmap_update_bits(wm8962->regmap, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + + /* Stereo control for EQ */ + regmap_update_bits(wm8962->regmap, WM8962_EQ1, + WM8962_EQ_SHARED_COEFF, 0); + + /* Don't debouce interrupts so we don't need SYSCLK */ + regmap_update_bits(wm8962->regmap, WM8962_IRQ_DEBOUNCE, + WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | + WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, + 0); + if (wm8962->pdata.in4_dc_measure) { ret = regmap_register_patch(wm8962->regmap, wm8962_dc_measure, -- cgit From e58f301ec969430cdafd7fa872660458f4939507 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Oct 2013 17:26:22 +0100 Subject: ASoC: rt5640: Power down LDO while suspended If we have control over the LDO then disable it during suspend; the device is already being put into reset so will be non-functional over suspend anyway and this will save a small amount of power. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/rt5640.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 641eeeb00c5c..b0cde92be7eb 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1979,12 +1979,20 @@ static int rt5640_suspend(struct snd_soc_codec *codec) rt5640_reset(codec); regcache_cache_only(rt5640->regmap, true); regcache_mark_dirty(rt5640->regmap); + if (gpio_is_valid(rt5640->pdata.ldo1_en)) + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 0); return 0; } static int rt5640_resume(struct snd_soc_codec *codec) { + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1); + msleep(400); + } rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit From ebce31140c2ddfda005e88957ac1ee1eacaa8dc5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Oct 2013 17:30:44 +0100 Subject: ASoC: rt5640: Don't go to standby on resume There is no need for the CODEC to go to standby on resume since the core will power it up as needed and in any case it is an idle_bias_off CODEC so would normally sit with bias off while idle. Signed-off-by: Mark Brown Tested-by: Stephen Warren --- sound/soc/codecs/rt5640.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b0cde92be7eb..4d041d376f31 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1993,7 +1993,6 @@ static int rt5640_resume(struct snd_soc_codec *codec) gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1); msleep(400); } - rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } -- cgit From 7b09eea52939d2b979f19de40e34b8670feff4c5 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 18 Oct 2013 14:30:01 -0500 Subject: ASoC: cs42l73: Add Device Tree support for CS42L73 This patch adds support for device tree for the CS42L73 CODEC Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 29 +++++++++++++++++++++++++++-- 1 file changed, 27 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 89efc3c6aefc..549d5d6a3fef 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,7 +17,7 @@ #include #include #include -#include +#include #include #include #include @@ -1416,6 +1416,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), GFP_KERNEL); @@ -1431,8 +1432,25 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return ret; } - if (pdata) + if (pdata) { cs42l73->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l73_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_u32(i2c_client->dev.of_node, + "chgfreq", &val32) >= 0) + pdata->chgfreq = val32; + } + pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, + "reset-gpio", 0); + cs42l73->pdata = *pdata; + } i2c_set_clientdata(i2c_client, cs42l73); @@ -1493,6 +1511,12 @@ static int cs42l73_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l73_of_match[] = { + { .compatible = "cirrus,cs42l73", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l73_of_match); + static const struct i2c_device_id cs42l73_id[] = { {"cs42l73", 0}, {} @@ -1504,6 +1528,7 @@ static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", .owner = THIS_MODULE, + .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, .probe = cs42l73_i2c_probe, -- cgit From cfcff69af8447df8dd3c5b14349c3b84b8b569a5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 20 Oct 2013 18:14:20 +0100 Subject: ASoC: si476x: Fix locking of core The conversion of the si476x to regmap removed locking of the core during register updates, allowing things like power state changes for the MFD to happen during a register update. Avoid this by taking the core lock in the DAI operations (which are the only things that do register updates) as we used to do in the open coded register I/O functions. Signed-off-by: Mark Brown Acked-by: Andrey Smirnov --- sound/soc/codecs/si476x.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 03645ce42063..52e7cb08434b 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -73,6 +73,7 @@ static const struct snd_soc_dapm_route si476x_dapm_routes[] = { static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { + struct si476x_core *core = i2c_mfd_cell_to_core(codec_dai->dev); int err; u16 format = 0; @@ -136,9 +137,14 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_update_bits(codec_dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK, format); + + si476x_core_unlock(core); + if (err < 0) { dev_err(codec_dai->codec->dev, "Failed to set output format\n"); return err; @@ -151,6 +157,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct si476x_core *core = i2c_mfd_cell_to_core(dai->dev); int rate, width, err; rate = params_rate(params); @@ -176,11 +183,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_write(dai->codec, SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE, rate); if (err < 0) { dev_err(dai->codec->dev, "Failed to set sample rate\n"); - return err; + goto out; } err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, @@ -189,10 +198,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)); if (err < 0) { dev_err(dai->codec->dev, "Failed to set output width\n"); - return err; + goto out; } - return 0; +out: + si476x_core_unlock(core); + + return err; } static int si476x_codec_probe(struct snd_soc_codec *codec) -- cgit From f95a48834cb9c581eec952215666a323136f339f Mon Sep 17 00:00:00 2001 From: Sebastian Reichel Date: Wed, 23 Oct 2013 14:03:28 +0200 Subject: ASoC: tpa6130a2: Add device tree support Add device tree support to tpa6130a2 driver and document the bindings. Signed-off-by: Sebastian Reichel Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 32 +++++++++++++++++++++++--------- 1 file changed, 23 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index c58bee8346ce..998555f2a8aa 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -30,6 +30,7 @@ #include #include #include +#include #include "tpa6130a2.h" @@ -364,30 +365,33 @@ static int tpa6130a2_probe(struct i2c_client *client, { struct device *dev; struct tpa6130a2_data *data; - struct tpa6130a2_platform_data *pdata; + struct tpa6130a2_platform_data *pdata = client->dev.platform_data; + struct device_node *np = client->dev.of_node; const char *regulator; int ret; dev = &client->dev; - if (client->dev.platform_data == NULL) { - dev_err(dev, "Platform data not set\n"); - dump_stack(); - return -ENODEV; - } - data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); if (data == NULL) { dev_err(dev, "Can not allocate memory\n"); return -ENOMEM; } + if (pdata) { + data->power_gpio = pdata->power_gpio; + } else if (np) { + data->power_gpio = of_get_named_gpio(np, "power-gpio", 0); + } else { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + tpa6130a2_client = client; i2c_set_clientdata(tpa6130a2_client, data); - pdata = client->dev.platform_data; - data->power_gpio = pdata->power_gpio; data->id = id->driver_data; mutex_init(&data->mutex); @@ -466,10 +470,20 @@ static const struct i2c_device_id tpa6130a2_id[] = { }; MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tpa6130a2_of_match[] = { + { .compatible = "ti,tpa6130a2", }, + { .compatible = "ti,tpa6140a2" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tpa6130a2_of_match); +#endif + static struct i2c_driver tpa6130a2_i2c_driver = { .driver = { .name = "tpa6130a2", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tpa6130a2_of_match), }, .probe = tpa6130a2_probe, .remove = tpa6130a2_remove, -- cgit From f4cdb6b493ec4cc97387e81a1e0c50335ed0b45d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 12 Nov 2013 10:52:10 +0000 Subject: ASoC: wm8997: Correct typo in ISRC mux routes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 6ec3de3efa4f..2e886caf44de 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -887,7 +887,7 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), -- cgit From 12850b8d45cb03239ede6df58ad0022aba3f3dc2 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 13 Nov 2013 13:01:49 +0000 Subject: ASoC: arizona: Fix typo in name of EQ coefficient controls Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 8 ++++---- sound/soc/codecs/wm5110.c | 8 ++++---- sound/soc/codecs/wm8997.c | 8 ++++---- 3 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8bbddc151aa8..a08e8bf6d07c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,13 +685,13 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, +SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, +SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, +SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, +SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, ARIZONA_EQ4_ENA_MASK), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd64384ca1c..b67ba33546d6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -101,13 +101,13 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, +SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, +SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, +SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, +SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, ARIZONA_EQ4_ENA_MASK), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 2e886caf44de..1392bb3c9254 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -170,13 +170,13 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, +SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, +SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, +SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, +SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, ARIZONA_EQ4_ENA_MASK), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, -- cgit