March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: wm8994: Manage wm8994's MCLK in codec driver
by Inha Song 16 Mar '15

16 Mar '15

Previously, When we use wm8994 codec, we should control its MCLK in machine driver. But, It should be managed by wm8994 codec driver, not machine driver. This patch add MCLK clock DT parsing logic and control own MCLK. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- sound/soc/codecs/wm8994.c | 22 +++++++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4fbc768..acb6b44 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -2471,6 +2472,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + int i; wm_hubs_set_bias_level(codec, level); @@ -2492,8 +2494,15 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) + clk_prepare_enable(wm8994->clk_mclk[i]); + active_reference(codec); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_ON) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) + clk_disable_unprepare(wm8994->clk_mclk[i]); + } break; case SND_SOC_BIAS_STANDBY: @@ -4449,6 +4458,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { static int wm8994_probe(struct platform_device *pdev) { struct wm8994_priv *wm8994; + int i; wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -4460,6 +4470,16 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + char mclk_name[] = "MCLKx"; + + snprintf(mclk_name, sizeof(mclk_name), "MCLK%d", i + 1); + + wm8994->clk_mclk[i] = devm_clk_get(pdev->dev.parent, mclk_name); + if (IS_ERR(wm8994->clk_mclk[i])) + wm8994->clk_mclk[i] = NULL; + } + pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index dd73387..d426c44 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { int sysclk[2]; int sysclk_rate[2]; int mclk[2]; + struct clk *clk_mclk[2]; int aifclk[2]; int aifdiv[2]; int channels[2]; -- 2.0.0.390.gcb682f8

2 1

[alsa-devel] [RFC PATCH] ASoC: wm8731: let codec to manage clock by itself
by Songjun Wu 16 Mar '15

16 Mar '15

Enable WM8731 to support common clock framework. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/codecs/wm8731.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 098c143..8df1550 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -25,6 +25,7 @@ #include <linux/spi/spi.h> #include <linux/of_device.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -390,6 +392,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (wm8731->mclk && clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -491,6 +495,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (wm8731->mclk) + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -509,6 +515,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (wm8731->mclk) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -667,6 +675,19 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&spi->dev, "Assuming static MCLK\n"); + } else { + dev_err(&spi->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -718,6 +739,19 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + if (ret == -ENOENT) { + wm8731->mclk = NULL; + dev_warn(&i2c->dev, "Assuming static MCLK\n"); + } else { + dev_err(&i2c->dev, "Failed to get MCLK: %d\n", + ret); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); -- 1.7.9.5

3 2

[alsa-devel] [PATCH 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index c7ffafa..a6e7afa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 2.1.4

3 3

[alsa-devel] [PATCH] ASoC: rt5677: add API to select ASRC clock source
by Oder Chiou 16 Mar '15

16 Mar '15

This patch defines an API to select the clock source for specified filters. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 163 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 79 ++++++++++++++++++++++ 2 files changed, 242 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c2a6e40..af18258 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1034,6 +1034,169 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } +/** + * rt5677_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5677 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc3_mask = 0, asrc3_value = 0; + unsigned int asrc4_mask = 0, asrc4_value = 0; + unsigned int asrc5_mask = 0, asrc5_value = 0; + unsigned int asrc6_mask = 0, asrc6_value = 0; + unsigned int asrc7_mask = 0, asrc7_value = 0; + + switch (clk_src) { + case RT5677_CLK_SEL_SYS: + case RT5677_CLK_SEL_I2S1_ASRC: + case RT5677_CLK_SEL_I2S2_ASRC: + case RT5677_CLK_SEL_I2S3_ASRC: + case RT5677_CLK_SEL_I2S4_ASRC: + case RT5677_CLK_SEL_I2S5_ASRC: + case RT5677_CLK_SEL_I2S6_ASRC: + case RT5677_CLK_SEL_SYS2: + case RT5677_CLK_SEL_SYS3: + case RT5677_CLK_SEL_SYS4: + case RT5677_CLK_SEL_SYS5: + case RT5677_CLK_SEL_SYS6: + case RT5677_CLK_SEL_SYS7: + break; + + default: + return -EINVAL; + } + + /* ASRC 3 */ + if (filter_mask & RT5677_DA_STEREO_FILTER) { + asrc3_mask |= RT5677_DA_STO_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5677_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_L_FILTER) { + asrc3_mask |= RT5677_DA_MONO2L_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO2_R_FILTER) { + asrc3_mask |= RT5677_DA_MONO2R_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5677_DA_MONO2R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO2R_CLK_SEL_SFT); + } + + if (asrc3_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_3, asrc3_mask, + asrc3_value); + + /* ASRC 4 */ + if (filter_mask & RT5677_DA_MONO3_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO3L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO3_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO3R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO3R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO3R_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_L_FILTER) { + asrc4_mask |= RT5677_DA_MONO4L_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4L_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4L_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DA_MONO4_R_FILTER) { + asrc4_mask |= RT5677_DA_MONO4R_CLK_SEL_MASK; + asrc4_value = (asrc4_value & ~RT5677_DA_MONO4R_CLK_SEL_MASK) + | (clk_src << RT5677_DA_MONO4R_CLK_SEL_SFT); + } + + if (asrc4_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_4, asrc4_mask, + asrc4_value); + + /* ASRC 5 */ + if (filter_mask & RT5677_AD_STEREO1_FILTER) { + asrc5_mask |= RT5677_AD_STO1_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO2_FILTER) { + asrc5_mask |= RT5677_AD_STO2_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO2_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO2_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO3_FILTER) { + asrc5_mask |= RT5677_AD_STO3_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO3_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_STEREO4_FILTER) { + asrc5_mask |= RT5677_AD_STO4_CLK_SEL_MASK; + asrc5_value = (asrc5_value & ~RT5677_AD_STO4_CLK_SEL_MASK) + | (clk_src << RT5677_AD_STO4_CLK_SEL_SFT); + } + + if (asrc5_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_5, asrc5_mask, + asrc5_value); + + /* ASRC 6 */ + if (filter_mask & RT5677_AD_MONO_L_FILTER) { + asrc6_mask |= RT5677_AD_MONOL_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_AD_MONO_R_FILTER) { + asrc6_mask |= RT5677_AD_MONOR_CLK_SEL_MASK; + asrc6_value = (asrc6_value & ~RT5677_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5677_AD_MONOR_CLK_SEL_SFT); + } + + if (asrc6_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_6, asrc6_mask, + asrc6_value); + + /* ASRC 7 */ + if (filter_mask & RT5677_DSP_OB_0_3_FILTER) { + asrc7_mask |= RT5677_DSP_OB_0_3_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_0_3_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_0_3_CLK_SEL_SFT); + } + + if (filter_mask & RT5677_DSP_OB_4_7_FILTER) { + asrc7_mask |= RT5677_DSP_OB_4_7_CLK_SEL_MASK; + asrc7_value = (asrc7_value & ~RT5677_DSP_OB_4_7_CLK_SEL_MASK) + | (clk_src << RT5677_DSP_OB_4_7_CLK_SEL_SFT); + } + + if (asrc7_mask) + regmap_update_bits(rt5677->regmap, RT5677_ASRC_7, asrc7_mask, + asrc7_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5677_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5677_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5677_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 07df96b..9dceb41 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1406,6 +1406,46 @@ #define RT5677_DSP_CLK_SRC_PLL2 (0x0 << 7) #define RT5677_DSP_CLK_SRC_BYPASS (0x1 << 7) +/* ASRC Control 3 (0x85) */ +#define RT5677_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_STO_CLK_SEL_SFT 12 +#define RT5677_DA_MONO2L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO2L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO2R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO2R_CLK_SEL_SFT 0 + +/* ASRC Control 4 (0x86) */ +#define RT5677_DA_MONO3L_CLK_SEL_MASK (0xf << 12) +#define RT5677_DA_MONO3L_CLK_SEL_SFT 12 +#define RT5677_DA_MONO3R_CLK_SEL_MASK (0xf << 8) +#define RT5677_DA_MONO3R_CLK_SEL_SFT 8 +#define RT5677_DA_MONO4L_CLK_SEL_MASK (0xf << 4) +#define RT5677_DA_MONO4L_CLK_SEL_SFT 4 +#define RT5677_DA_MONO4R_CLK_SEL_MASK (0xf << 0) +#define RT5677_DA_MONO4R_CLK_SEL_SFT 0 + +/* ASRC Control 5 (0x87) */ +#define RT5677_AD_STO1_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_STO1_CLK_SEL_SFT 12 +#define RT5677_AD_STO2_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_STO2_CLK_SEL_SFT 8 +#define RT5677_AD_STO3_CLK_SEL_MASK (0xf << 4) +#define RT5677_AD_STO3_CLK_SEL_SFT 4 +#define RT5677_AD_STO4_CLK_SEL_MASK (0xf << 0) +#define RT5677_AD_STO4_CLK_SEL_SFT 0 + +/* ASRC Control 6 (0x88) */ +#define RT5677_AD_MONOL_CLK_SEL_MASK (0xf << 12) +#define RT5677_AD_MONOL_CLK_SEL_SFT 12 +#define RT5677_AD_MONOR_CLK_SEL_MASK (0xf << 8) +#define RT5677_AD_MONOR_CLK_SEL_SFT 8 + +/* ASRC Control 7 (0x89) */ +#define RT5677_DSP_OB_0_3_CLK_SEL_MASK (0xf << 12) +#define RT5677_DSP_OB_0_3_CLK_SEL_SFT 12 +#define RT5677_DSP_OB_4_7_CLK_SEL_MASK (0xf << 8) +#define RT5677_DSP_OB_4_7_CLK_SEL_SFT 8 + /* VAD Function Control 4 (0x9f) */ #define RT5677_VAD_SRC_MASK (0x7 << 8) #define RT5677_VAD_SRC_SFT 8 @@ -1670,6 +1710,42 @@ enum rt5677_type { RT5676, }; +/* ASRC clock source selection */ +enum { + RT5677_CLK_SEL_SYS, + RT5677_CLK_SEL_I2S1_ASRC, + RT5677_CLK_SEL_I2S2_ASRC, + RT5677_CLK_SEL_I2S3_ASRC, + RT5677_CLK_SEL_I2S4_ASRC, + RT5677_CLK_SEL_I2S5_ASRC, + RT5677_CLK_SEL_I2S6_ASRC, + RT5677_CLK_SEL_SYS2, + RT5677_CLK_SEL_SYS3, + RT5677_CLK_SEL_SYS4, + RT5677_CLK_SEL_SYS5, + RT5677_CLK_SEL_SYS6, + RT5677_CLK_SEL_SYS7, +}; + +/* filter mask */ +enum { + RT5677_DA_STEREO_FILTER = 0x1, + RT5677_DA_MONO2_L_FILTER = (0x1 << 1), + RT5677_DA_MONO2_R_FILTER = (0x1 << 2), + RT5677_DA_MONO3_L_FILTER = (0x1 << 3), + RT5677_DA_MONO3_R_FILTER = (0x1 << 4), + RT5677_DA_MONO4_L_FILTER = (0x1 << 5), + RT5677_DA_MONO4_R_FILTER = (0x1 << 6), + RT5677_AD_STEREO1_FILTER = (0x1 << 7), + RT5677_AD_STEREO2_FILTER = (0x1 << 8), + RT5677_AD_STEREO3_FILTER = (0x1 << 9), + RT5677_AD_STEREO4_FILTER = (0x1 << 10), + RT5677_AD_MONO_L_FILTER = (0x1 << 11), + RT5677_AD_MONO_R_FILTER = (0x1 << 12), + RT5677_DSP_OB_0_3_FILTER = (0x1 << 13), + RT5677_DSP_OB_4_7_FILTER = (0x1 << 14), +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1696,4 +1772,7 @@ struct rt5677_priv { bool is_vref_slow; }; +int rt5677_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: nuc900: No need to track the dma buffer in the driver state struct
by Lars-Peter Clausen 16 Mar '15

16 Mar '15

The DMA buffer and address can be accessed through the snd_pcm_runtime. There is no need to manually track them in the driver's state struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/nuc900/nuc900-audio.h | 3 --- sound/soc/nuc900/nuc900-pcm.c | 31 +++++-------------------------- 2 files changed, 5 insertions(+), 29 deletions(-) diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h index 59f7e8e..d0b7257 100644 --- a/sound/soc/nuc900/nuc900-audio.h +++ b/sound/soc/nuc900/nuc900-audio.h @@ -100,10 +100,7 @@ struct nuc900_audio { void __iomem *mmio; spinlock_t lock; - dma_addr_t dma_addr[2]; - unsigned long buffersize[2]; unsigned long irq_num; - struct snd_pcm_substream *substream; struct resource *res; struct clk *clk; struct device *dev; diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index b809fa9..5ae5ca1 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -42,29 +42,10 @@ static const struct snd_pcm_hardware nuc900_pcm_hardware = { static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct nuc900_audio *nuc900_audio = runtime->private_data; - unsigned long flags; - int ret = 0; - - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; - - spin_lock_irqsave(&nuc900_audio->lock, flags); - - nuc900_audio->substream = substream; - nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; - nuc900_audio->buffersize[substream->stream] = - params_buffer_bytes(params); - - spin_unlock_irqrestore(&nuc900_audio->lock, flags); - - return ret; + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static void nuc900_update_dma_register(struct snd_pcm_substream *substream, - dma_addr_t dma_addr, size_t count) +static void nuc900_update_dma_register(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct nuc900_audio *nuc900_audio = runtime->private_data; @@ -78,8 +59,8 @@ static void nuc900_update_dma_register(struct snd_pcm_substream *substream, mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; } - AUDIO_WRITE(mmio_addr, dma_addr); - AUDIO_WRITE(mmio_len, count); + AUDIO_WRITE(mmio_addr, runtime->dma_addr); + AUDIO_WRITE(mmio_len, runtime->dma_bytes); } static void nuc900_dma_start(struct snd_pcm_substream *substream) @@ -170,9 +151,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream) spin_lock_irqsave(&nuc900_audio->lock, flags); - nuc900_update_dma_register(substream, - nuc900_audio->dma_addr[substream->stream], - nuc900_audio->buffersize[substream->stream]); + nuc900_update_dma_register(substream); val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); -- 1.8.0

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[alsa-devel] [PATCH] ASoC: Make snd_soc_dapm_kcontrol_codec() inline
by Lars-Peter Clausen 16 Mar '15

16 Mar '15

snd_soc_dapm_kcontrol_codec() is a extremely simple function and inlining it typically results in less code than necessary for calling the non-inlined version of the function. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-dapm.c | 10 ---------- 3 files changed, 13 insertions(+), 11 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8d7416e..78633ef 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -440,7 +440,6 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list); -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/include/sound/soc.h b/include/sound/soc.h index bc6cfab..92b92ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1268,6 +1268,19 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm( return component->dapm_ptr; } +/** + * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol + * @kcontrol: The kcontrol + * + * This function must only be used on DAPM contexts that are known to be part of + * a CODEC (e.g. in a CODEC driver). Otherwise the behavior is undefined. + */ +static inline struct snd_soc_codec *snd_soc_dapm_kcontrol_codec( + struct snd_kcontrol *kcontrol) +{ + return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); +} + /* codec IO */ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6f8820..95337c8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -473,16 +473,6 @@ struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( } EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_dapm); -/** - * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol - * @kcontrol: The kcontrol - */ -struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol) -{ - return snd_soc_dapm_to_codec(snd_soc_dapm_kcontrol_dapm(kcontrol)); -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_codec); - static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; -- 1.8.0

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[alsa-devel] [PATCH V8 0/5] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 16 Mar '15

16 Mar '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This patch series adds support for I2S audio playback on the Storm board, which contains a Qualcomm Technologies ipq806x SOC and a Maxim max98357a DAC/amp. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This contains an MI2S port, which is what these drivers are configured to use. The LPAIF also contains a DMA engine that is dedicated to moving audio samples into the transmit FIFO of the MI2S port. One bus from the MI2S port of the SOC is connected to the DAC/amp for stereo playback. This bus is configured so that the SOC is bus master and consists of DATA, LRCLK, and BCLK. The DAC/amp does not need MCLK to operate. In addition, a single GPIO pin from the SOC is connected to the same DAC/amp, which gives enable/disable control over the DAC/amp. The specific drivers added are: * a codec DAI driver for controlling the DAC/amp (applied from V4) * a CPU DAI driver for controlling the MI2S port (applied from V7) * a platform driver for controlling the LPAIF DMA engine (applied from V7) * a machine driver that instantiates a dai-link for playback The LPASS also contains clocks that need to be controlled. Those drivers have been submitted as several separate patch series: * [PATCH v3 0/8] qcom audio clock control drivers http://lkml.org/lkml/2015/1/19/656 * [PATCH] arm: dts: qcom: Add LCC nodes https://lkml.org/lkml/2015/1/28/557 * [PATCH 1/2] arm: qcom_defconfig: Enable lpass clock driver https://lkml.org/lkml/2015/1/28/582 * [PATCH] clk: qcom: Fix ipq806x LCC frequency tables https://lkml.org/lkml/2015/2/26/774 * [PATCH v2] clk: qcom: Properly change rates for ahbix clock https://lkml.org/lkml/2015/3/6/878 Even though the ipq806x LPASS does not contain an audio DSP, other SOCs do have one. For those SOCs, the audio DSP typically controls the hardware blocks in the LPASS. Hence, different CPU DAI driver(s) would need to be used in order to facilitate audio with the DSP. As such, the LPASS DT binding description alludes to an optional adsp phandle, which is not present for this SOC. For other SOCs that do contain an audio DSP, the phandle should exist and refer to a an adsp node. Not using the audio DSP on an SOC that does have one would require different CPU DAI driver(s), in addition to possible bootloader and/or firmware changes. Note that the use of a phandle was a modification that has been submitted as part of a separate patch series. Corresponding additions to the device tree for the ipq806x SOC and its documentation have also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well (applied from V7). = Changes since V7 [Patch V7 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/088546.html * Changed the compatible property of the machine driver to make it specific to audio * Removed the lcc node definition from the DT, as it was submitted separately = Changes since V6 [Patch V6 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/088218.h… * Added REGMAP_MMIO selection for the CPU and platform drivers * Modified the AHBIX frequency to match the corresponding LCC fix * Tweaked the logging in the CPU driver probe. = Changes since V5 [Patch V5 00/12] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087832.h… * Correctly use Storm as the build target label and DT binding label. * Added audio DSP sub-node to the LPASS device tree, disabled for this SOC. * Added logic to CPU DAI driver to fail the probe() if a DSP is present. * Use the standard naming convention for the DAI link properties. * General code cleanup. = Changes since V4 [Patch V4 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087499.h… * Replaced simple-card with a machine driver to resolve the system clock configuration, rather than having the CPU DAI driver do it. * Added header files to avoid indirect header dependencies and implicit forward declarations. * Tweaked the ISR to match the conventions of the surrounding code. * Removed the usage of the low-power memory as it is not needed. * Removed the use of the DRV_NAME constant. * Added explicit dependency on gpiolib for the codec driver. * Moved the MODULE_DEVICE_TABLE macro inside the CONFIG_OF conditional. * Modified the documentation to reflect the changes. * General code cleanup. = Changes since V3 [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085694.h… * Placed the content of the inline functions into the callbacks. * Replaced use of readl/writel register access functions with regmap access functions. Notable exception is the ISR, which uses ioread32/iowrite32. * Rearranged the sequencing of the hardware block enables to fit within the ASoC framework callbacks, while remaining functional. REQ 1: The hardware requires the enable sequence to be: LPAIF-DMA[enable],then LPAIF-MI2S[enable], then DAC-GPIO[enable] REQ 2: The hardware requires the disable sequence to be: DAC-GPIO[disable], then LPAIF-MI2S[disable] * Corrected the implementation of the pointer callback. * Utilize the LPM to buffer audio samples, rather than memory external to LPASS. * Corrected the interrupt clearing in the ISR. * Implemented a default system clock (defined by the simple-card DT node), and optional LPASS DT node modifiers that can alter the system clock in order to expand the range of available bit clock frequencies. * Addressed all of the remaining issues raised by Mark Brown. * General code cleanup. = Changes since V2 [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-December/085186.h… * Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. * Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. * Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. * Removed machine driver and tied DAI drivers to simple-audio-card. * Packaged the build files and Maxim codec files together in one change. * Removed QCOM as vendor from Maxim code and documentation. * Separated the SOC and board definitions into the correct DTS files. * Update device tree documentation to reflect changes. * General code cleanup. = Changes since V1 [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084322.h… * Remove the native LPAIF driver, and move its functionality to the CPU DAI driver. * Add a codec driver to manage the pins going to the external DAC (previously managed by the machine driver). * Use devm_* and dev_* where possible. * ISR only handles relevant DMA channel now. * Update device tree documentation to reflect changes. * General code cleanup. Kenneth Westfield (5): ASoC: qcom: Document Storm bindings ASoC: qcom: Add Storm machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware Documentation/devicetree/bindings/sound/storm.txt | 23 +++ arch/arm/boot/dts/qcom-ipq8064.dtsi | 16 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/qcom/Kconfig | 25 ++++ sound/soc/qcom/Makefile | 11 ++ sound/soc/qcom/storm.c | 162 ++++++++++++++++++++++ 7 files changed, 239 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/storm.txt create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/storm.c -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH 1/2] ASoC: qcom: Change qcom, adsp in LPASS CPU bindings
by Kenneth Westfield 16 Mar '15

16 Mar '15

From: Kenneth Westfield <kwestfie(a)codeaurora.org> Change the representation of the audio DSP, in the LPASS CPU bindings description, from a required subnode to an optional phandle. Signed-off-by: Kenneth Westfield <kwestfie(a)codeaurora.org> --- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index e7c6e9321863f022ebf0d51b75d7bb83c10b9062..e00732dac9393e95f0530505c9f38127f58980a3 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -22,13 +22,9 @@ Required properties: - reg-names : A list which must include the following entries: * "lpass-lpaif" -Required subnodes: +Optional properties: -- qcom,adsp : Audio DSP sub-node - -Optional Audio DSP subnode properties: - -- status : "disabled" indicates the adsp is not available. +- qcom,adsp : Phandle for the audio DSP node Example: @@ -43,7 +39,5 @@ lpass@28100000 { pinctrl-1 = <&mi2s_idle>; reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; - qcom,adsp { - status = "disabled"; - }; + qcom,adsp = <&adsp>; }; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH] ALSA: hda - Don't access stereo amps for mono channel widgets
by Takashi Iwai 16 Mar '15

16 Mar '15

The current HDA generic parser initializes / modifies the amp values always in stereo, but this seems causing the problem on ALC3229 codec that has a few mono channel widgets: namely, these mono widgets react to actions for both channels equally. In the driver code, we do care the mono channel and create a control only for the left channel (as defined in HD-audio spec) for such a node. When the control is updated, only the left channel value is changed. However, in the resume, the right channel value is also restored from the initial value we took as stereo, and this overwrites the left channel value. This ends up being the silent output as the right channel has been never touched and remains muted. This patch covers the places where unconditional stereo amp accesses are done and converts to the conditional accesses. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 30 ++++++++++++++++++++++-------- 1 file changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..fe18071bf93a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { unsigned int caps = query_amp_caps(codec, nid, dir); int val = get_amp_val_to_activate(codec, nid, dir, caps, false); - snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val); + else + snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val); +} + +/* update the amp, doing in stereo or mono depending on NID */ +static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, + unsigned int mask, unsigned int val) +{ + if (get_wcaps(codec, nid) & AC_WCAP_STEREO) + return snd_hda_codec_amp_stereo(codec, nid, dir, idx, + mask, val); + else + return snd_hda_codec_amp_update(codec, nid, 0, dir, idx, + mask, val); } /* calculate amp value mask we can modify; @@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, return; val &= mask; - snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val); + update_amp(codec, nid, dir, idx, mask, val); } static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, @@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) has_amp = nid_has_mute(codec, mix, HDA_INPUT); for (i = 0; i < nums; i++) { if (has_amp) - snd_hda_codec_amp_stereo(codec, mix, - HDA_INPUT, i, - 0xff, HDA_AMP_MUTE); + update_amp(codec, mix, HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) - snd_hda_codec_amp_stereo(codec, conn[i], - HDA_OUTPUT, 0, - 0xff, HDA_AMP_MUTE); + update_amp(codec, conn[i], HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); } } -- 2.3.2

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[alsa-devel] [PATCH 0/13] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 16 Mar '15

16 Mar '15

Hi Mark These are DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. Current ASoC DPCM needs route settings for FE/BE, but sometimes it will have trouble, because many drivers are using same name for this routing. This patch set solved this issue too. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : DAPM routing issue solution patches 5 - 6 : new Renesas sound card driver 7 - 11 : fixup/tidyup patches for rsnd 12 - 13 : DPCM support patch for rsnd Kuninori Morimoto (13): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: dapm: add snd_soc_dapm_route_scan macro for DAPM route 4) ASoC: dapm: add driver prefix matching for snd_soc_dapm_add_route() 5) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 6) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 7) ASoC: rsnd: no more SSI restart when unusual situation 8) ASoC: rsnd: no more SRC restart when unusual situation 9) ASoC: rsnd: tidyup error message format 10) ASoC: rsnd: show debug info for sampling rate convert 11) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 12) ASoC: rsnd: add dai_link stream name 13) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 491 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ sound/soc/soc-dapm.c | 33 +++++-- 13 files changed, 780 insertions(+), 75 deletions(-) Best regards --- Kuninori Morimoto

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