July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH] ASoC: codecs: add codec driver for MFD mc13783
by Jürgen Lambrecht 14 Jul '11

14 Jul '11

Sascha Hauer's driver from Pengutronix' imx-sound branch. Added some code found in a linux 2.6.33. Needs further work and cleanup. Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 29 ++ 4 files changed, 816 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..77fb7b2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if SPI select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -200,6 +201,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_MC13783 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..a129af2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 0000000..af7603a --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,781 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer(a)pengutronix.de + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/device.h> +#include <linux/mfd/mc13783.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define PMIC_AUDIO_RX_0 36 +#define PMIC_AUDIO_RX_1 37 +#define PMIC_AUDIO_TX 38 +#define PMIC_SSI_NETWORK 39 +#define PMIC_AUDIO_CODEC 40 +#define PMIC_AUDIO_DAC 41 + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_STDCSLOT(x) (((x) & 0x3) << 12) +#define SSI_NETWORK_STDCRXSLOT(x) (((x) & 0x3) << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +#define STEREO_DAC_STD_SSI_SEL (1 << 0) +#define STEREO_DAC_STD_CLK_SEL (1 << 1) +#define STEREO_DAC_STD_CSM (1 << 2) +#define STEREO_DAC_STD_BCL_INV (1 << 3) +#define STEREO_DAC_STD_CFS_INV (1 << 4) +#define STEREO_DAC_STD_CFS(x) (((x) & 0x3) << 5) +#define STEREO_DAC_STD_CLK(x) (((x) & 0x7) << 7) +#define STEREO_DAC_STD_CFS_DLY_B (1 << 10) +#define STEREO_DAC_STD_C_EN (1 << 11) +#define STEREO_DAC_STD_C_CLK_EN (1 << 12) +#define STEREO_DAC_STD_C_RESET (1 << 15) +#define STEREO_DAC_SPDIF (1 << 16) +#define STEREO_DAC_SR(x) (((x) & 0xf) << 17) + +#define RX_0_VAUDIOON 0 +#define RX_0_BIASEN 1 +#define RX_0_BIASSPEED 2 +#define RX_0_ASPEN 3 +#define RX_0_ASPSEL 4 +#define RX_0_ALSPEN 5 +#define RX_0_ALSPREF 6 +#define RX_0_ALSPSEL 7 +#define RX_0_LSPLEN 8 +#define RX_0_AHSREN 9 +#define RX_0_AHSLEN 10 +#define RX_0_AHSSEL 11 +#define RX_0_HSPGDIG 12 +#define RX_0_HSDETEN 13 +#define RX_0_HSDETAUTOB 14 +#define RX_0_ARXOUTREN 15 +#define RX_0_ARXOUTLEN 16 +#define RX_0_ARXOUTSEL 17 +#define RX_0_CDCOUTEN 18 +#define RX_0_HSLDETEN 19 +#define RX_0_ADDCDC 21 +#define RX_0_ADDSTDC 22 +#define RX_0_ADDRXIN 23 + +#define RX_1_PGASTEN 5 +#define RX_1_PGAST 6 +#define RX_1_ARXINEN 10 +#define RX_1_ARXIN 11 +#define RX_1_PGARXIN 12 + +#define MC13783_REG_CACHE_SIZE 42 + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13783 *mc13783; + + u32 reg_cache[MC13783_REG_CACHE_SIZE]; + + int mc13783_asp_val; + int mc13783_alsp_val; + int mc13783_ahsx_val; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u32 ret; + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + if (reg >= MC13783_REG_CACHE_SIZE) { + ret = 0; + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + } else { + ret = priv->reg_cache[reg]; + } + return ret; +} + +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; + + if (reg >= MC13783_REG_CACHE_SIZE) { + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + ret = -1; + } else { + mc13783_lock(priv->mc13783); + priv->reg_cache[reg] = value; + + ret = mc13783_reg_write(priv->mc13783, reg, value); + + mc13783_unlock(priv->mc13783); + } + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + int i; + unsigned int reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, PMIC_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + } else { + if (rate == 8000) + reg = 0; + else + reg = (1 << 10); + + snd_soc_update_bits(codec, PMIC_AUDIO_CODEC, + 0x1 << 10, reg); + + return 0; + } + + return -EINVAL; +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CFS(3); + reg &= ~STEREO_DAC_STD_BCL_INV; + reg &= ~STEREO_DAC_STD_CFS_INV; + reg &= ~STEREO_DAC_STD_CSM; + reg &= ~STEREO_DAC_STD_C_CLK_EN; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= STEREO_DAC_STD_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= STEREO_DAC_STD_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + reg |= STEREO_DAC_STD_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= STEREO_DAC_STD_BCL_INV; + reg |= STEREO_DAC_STD_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + reg |= STEREO_DAC_STD_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= STEREO_DAC_STD_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= STEREO_DAC_STD_CSM; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + reg |= STEREO_DAC_STD_C_EN; /* DAC power up */ + reg |= STEREO_DAC_STD_C_RESET; /* reset filter */ + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CLK(0x7); + reg &= ~STEREO_DAC_STD_CLK_SEL; + + switch (freq) { + case 13000000: + clk = 0; + break; + case 15360000: + clk = 1; + break; + case 16800000: + clk = 2; + break; + case 26000000: + clk = 4; + break; + case 12000000: + clk = 5; + break; + case 3686400: + clk = 6; + break; + case 33600000: + clk = 7; + break; + default: + return -EINVAL; + } + + if (clk_id == MC13783_CLK_CLIB) + reg |= STEREO_DAC_STD_CLK_SEL; + + reg |= STEREO_DAC_STD_CLK(clk); + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (slots != 4) + return -EINVAL; + + reg = mc13783_read(codec, PMIC_SSI_NETWORK); + + reg &= ~(0xfff << 0); + reg |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + reg |= (0x01 << 4); /* secondary timeslot TX is 1 */ + reg |= (0x01 << 6); /* secondary timeslot RX is 1 */ + + mc13783_write(codec, PMIC_SSI_NETWORK, reg); + + return 0; +} + +static int mc13783_get_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_alsp_val; + + return 0; +} + +static int mc13783_put_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_alsp_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_ALSPEN) | (1 << RX_0_ALSPSEL)); + + if (priv->mc13783_alsp_val) + reg |= 1 << RX_0_ALSPEN; + + if (priv->mc13783_alsp_val == 2) + reg |= 1 << RX_0_ALSPSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +static int mc13783_get_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_ahsx_val; + + return 0; +} + +static int mc13783_put_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_ahsx_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN) | (1 << RX_0_AHSSEL)); + + if (priv->mc13783_ahsx_val) + reg |= (1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN); + + if (priv->mc13783_ahsx_val == 2) + reg |= 1 << RX_0_AHSSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; + } + +static int mc13783_pcm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDSTDC) & 1; + printk(KERN_DEBUG "pcm_get %d\n", ucontrol->value.enumerated.item[0]); + + return 0; +} + +static int mc13783_pcm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + + if (ucontrol->value.enumerated.item[0]) { + printk(KERN_DEBUG "pcm_put on %d\n", ucontrol->value.enumerated.item[0]); + r36 |= (1 << RX_0_ADDSTDC); + r37 |= (1 << RX_1_PGASTEN); + } else { + printk(KERN_DEBUG "pcm_put off\n"); + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_linein_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDRXIN) & 1; + + return 0; +} + +static int mc13783_linein_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + + if (ucontrol->value.enumerated.item[0]) { + r36 |= (1 << RX_0_ADDRXIN); + r37 |= (1 << RX_1_ARXINEN); + } else { + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_capure_cache; + +static int mc13783_get_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = mc13783_capure_cache; + return 0; +} + +#define AMC1REN (1 << 5) +#define AMC2EN (1 << 9) +#define ATXINEN (1 << 11) +#define RXINREC (1 << 13) +#define AMC1LEN (1 << 7) + +static int mc13783_put_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r38, change; + + r38 = mc13783_read(codec, PMIC_AUDIO_TX); + + change = (mc13783_capure_cache != ucontrol->value.enumerated.item[0]); + mc13783_capure_cache = ucontrol->value.enumerated.item[0]; + r38 &= ~(AMC1REN | AMC2EN | ATXINEN | RXINREC | AMC1LEN); + + switch (mc13783_capure_cache) { + case 0: + break; + case 1: + r38 |= RXINREC; + break; + case 2: + r38 |= AMC1REN | AMC1LEN; + break; + case 3: + r38 |= AMC1REN; + break; + case 4: + r38 |= AMC2EN; + break; + case 5: + r38 |= AMC1LEN | AMC2EN; + break; + case 6: + r38 |= ATXINEN; + break; + case 7: + r38 |= AMC1LEN | ATXINEN; + break; + case 8: + r38 |= AMC1LEN | RXINREC; + break; + case 9: + r38 |= AMC1LEN; + break; + default: + break; + } + + mc13783_write(codec, PMIC_AUDIO_TX, r38); + + return change; +} + +static const char *mc13783_asp[] = {"Off", "Codec", "Right"}; +static const char *mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_ahs[] = {"Codec", "Mixer"}; + +//static const char *mc13783_ahsx[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_arxout[] = {"Codec", "Mixer"}; + +static const char *mc13783_capture[] = {"off/off", "rxinl/rxinr", + "mc1lin/mc1rin", "off/mc1rin", "off/mc2in", "mc1lin/mc2in", + "off/txin", "mc1lin/txin", "mc1lin/rxinr", "mc1lin/off"}; + +static const char *mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +//static const char *mc13783_pcms[] = {"Off", "On"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 3, ARRAY_SIZE(mc13783_asp), mc13783_asp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_alsp), mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 17, ARRAY_SIZE(mc13783_arxout), + mc13783_arxout); + +static const struct soc_enum mc13783_enum_capture = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_capture), mc13783_capture); + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +//static const struct soc_enum mc13783_enum_ahsx = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 9, ARRAY_SIZE(mc13783_ahsx), +// mc13783_ahsx); +// +//static const struct soc_enum mc13783_enum_pcms = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 5, ARRAY_SIZE(mc13783_pcms), +// mc13783_pcms); + +static struct snd_kcontrol_new mc13783_control_list[] = { + /* Output Routing */ + SOC_ENUM("Asp Source", mc13783_enum_asp), + SOC_ENUM_EXT("Alsp Source", mc13783_enum_alsp, mc13783_get_alsp, + mc13783_put_alsp), + SOC_ENUM("Ahs Source", mc13783_enum_ahs), + SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 1, 0), + SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 1, 0), +/* instead of the above 3 lines, could put this line: */ +// SOC_ENUM_EXT("Ahs Source", mc13783_enum_ahsx, mc13783_get_ahsx, +// mc13783_put_ahsx), +//// SOC_ENUM("Ahs Source", mc13783_enum_ahs), +//// SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 2, 0), +//// SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 2, 0), +//// SOC_ENUM("Ahsr enable", mc13783_enum_ahsr), +//// SOC_ENUM("Ahsl enable", mc13783_enum_ahsl), + SOC_ENUM("Arxout Source", mc13783_enum_arxout), + SOC_SINGLE("ArxoutR enable", PMIC_AUDIO_RX_0, 16, 1, 0), + SOC_SINGLE("ArxoutL enable", PMIC_AUDIO_RX_0, 15, 1, 0), + SOC_SINGLE_EXT("PCM Playback Switch", 0, 0, 1, 0, mc13783_pcm_get, +// SOC_ENUM_EXT("PCMS Playback Switch", mc13783_enum_pcms, mc13783_pcm_get, + mc13783_pcm_put), + SOC_SINGLE("PCM Playback Volume", PMIC_AUDIO_RX_1, 6, 15, 0), +// SOC_SINGLE("PCMP Playback Volume", PMIC_AUDIO_RX_1, RX_1_PGAST, 15, 0), + SOC_SINGLE_EXT("Line in Switch", 0, 0, 1, 0, mc13783_linein_get, + mc13783_linein_put), + SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, 12, 15, 0), +// SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, RX_1_PGARXIN, 15, 0), + SOC_ENUM_EXT("Capture Source", mc13783_enum_capture, mc13783_get_capture, + mc13783_put_capture), + SOC_DOUBLE("PCM Capture Volume", PMIC_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + /* these are the reset values */ + priv->reg_cache[PMIC_AUDIO_RX_0] = (1 << RX_0_HSPGDIG); //0x001000; + priv->reg_cache[PMIC_AUDIO_RX_1] = 0x00d35A; + priv->reg_cache[PMIC_AUDIO_TX] = 0x420000; + priv->reg_cache[PMIC_SSI_NETWORK] = 0x013060; + priv->reg_cache[PMIC_AUDIO_CODEC] = 0x180027; + priv->reg_cache[PMIC_AUDIO_DAC] = 0x0e0004; + + /* VAUDIOON -> supply audio part, BIAS enable */ + priv->reg_cache[PMIC_AUDIO_RX_0] |= (1 << RX_0_VAUDIOON) | (1 << RX_0_BIASEN); //0x3; + + for (i = 36; i < 42; i++) + mc13783_write(codec, i, priv->reg_cache[i]); + + snd_soc_add_controls(codec, mc13783_control_list, + ARRAY_SIZE(mc13783_control_list)); + + return ret; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + unsigned int reg; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + /* VAUDIOON -> switch off audio part, BIAS disable */ + reg &= ~0x3; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) +/*Other functions (soc_dai.h): + * set_pll, set_clkdiv + * set_tdm_slot, set_channel_map, set_tristate + * digital_mute + * startup, shutdown, hw_free, prepare, trigger +*/ +static struct snd_soc_dai_ops mc13783_ops = { + .hw_params = mc13783_pcm_hw_params, + .set_fmt = mc13783_set_fmt, + .set_sysclk = mc13783_set_sysclk, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +static struct snd_soc_dai_driver mc13783_dai = { + .name = "mc13783-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops, +}; + +/* + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, +// .suspend +// .resume + .read = mc13783_read, + .write = mc13783_write, +// .display_register +// .volatile_register +// .readable_register +// .reg_cache_size = MC13783_REG_CACHE_SIZE, TODO: use codec->reg_cache instead of privately +// reg_cache_step +// .reg_word_size = sizeof(u32), TODO: else: kmemdup error in soc-core.c::snd_soc_register_codec +// .reg_cache_default = wm8960_reg, +// .compress_type +// .set_bias_level +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13783 *mc13783; + struct mc13783_priv *priv; + int ret; + + mc13783 = dev_get_drvdata(pdev->dev.parent); + + priv = kzalloc(sizeof(struct mc13783_priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13783 = mc13783; + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, &mc13783_dai, 1); + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + kfree(priv); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +static __init int mc13783_init(void) +{ + return platform_driver_register(&mc13783_codec_driver); +} + +static __exit void mc13783_exit(void) +{ + platform_driver_unregister(&mc13783_codec_driver); +} + +module_init(mc13783_init); +module_exit(mc13783_exit); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 0000000..7285370 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,29 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +extern int mc13783_add_ctl(struct snd_card*, void *); + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 + +#endif /* MC13783_MIXER_H */ -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Reduce power consumption for idle DAIs in WM8994
by Mark Brown 14 Jul '11

14 Jul '11

If DAIs are idle but their clocks are in use for some reason (eg, as SYSCLK or for accessory detect) then set the clock dividers to the maximum to reduce slightly the power consumption of the unclocked circuits. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 29 +++++++++++++++++++++++++++++ 1 files changed, 29 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 31854dc..bc62b6e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2311,6 +2311,33 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } +static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int rate_reg = 0; + + switch (dai->id) { + case 1: + rate_reg = WM8994_AIF1_RATE; + break; + case 2: + rate_reg = WM8994_AIF1_RATE; + break; + default: + break; + } + + /* If the DAI is idle then configure the divider tree for the + * lowest output rate to save a little power if the clock is + * still active (eg, because it is system clock). + */ + if (rate_reg && !dai->playback_active && !dai->capture_active) + snd_soc_update_bits(codec, rate_reg, + WM8994_AIF1_SR_MASK | + WM8994_AIF1CLK_RATE_MASK, 0x9); +} + static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2377,6 +2404,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2386,6 +2414,7 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
by Mark Brown 14 Jul '11

14 Jul '11

On Thu, Jul 14, 2011 at 06:14:46PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks

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[alsa-devel] www.alsa-project.org down?
by David Henningsson 14 Jul '11

14 Jul '11

Hi, Is www.alsa-project.org down? Someone in a bug report he couldn't connect to it and here's what happens when I try: $ wget http://www.alsa-project.org --2011-07-08 11:41:36-- http://www.alsa-project.org/ Resolving www.alsa-project.org... 77.48.224.243 Connecting to www.alsa-project.org|77.48.224.243|:80... connected. HTTP request sent, awaiting response... (It's been awaiting response for a few minutes now...) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] Specifying the default buffer-size for snd-hda-intel
by Raymond Yau 14 Jul '11

14 Jul '11

> +config SND_HDA_PREALLOC_SIZE > + int "Pre-allocated buffer size for HD-audio driver" > + range 0 32768 > + default 64 > + help > + Speficies the default pre-allocated buffer-size in kB for > + HD-audio driver. A larger buffer (e.g. 2048) is preferred > + for systems with PulseAudio. The default 64 is chosen just > + from the compatibility reason. > + + range 0 32768 - range 64 32768 It seem that the minimum range cannot be zero Does current PA server still work with sound drivers with 64K buffer ?

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[alsa-devel] [PATCH] ASoC: Log WM8994 FIFO errors from the interrupt
by Mark Brown 14 Jul '11

14 Jul '11

We should spot them anyway on state changes but logging them gives us better time information about when the misconfiguration happened. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index fcd067f..d774fd7a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2894,6 +2894,15 @@ out: return IRQ_HANDLED; } +static irqreturn_t wm8994_fifo_error(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "FIFO error\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2973,6 +2982,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, + wm8994_fifo_error, "FIFO error", codec); + ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", &wm8994->hubs); @@ -3255,6 +3267,7 @@ err_irq: &wm8994->fll_locked[i]); wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); err: kfree(wm8994); return ret; @@ -3276,6 +3289,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); switch (control->type) { case WM8994: -- 1.7.5.4

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 14 Jul '11

14 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (9): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/sta32x.c | 140 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic26.c | 14 +++- sound/soc/codecs/tlv320aic3x.c | 9 ++- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 101 +++++++++++++++++++++++----- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 37 ++++++++-- sound/soc/codecs/wm_hubs.h | 8 ++ sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 ++-- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- 13 files changed, 301 insertions(+), 45 deletions(-)

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Re: [alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Raymond Yau 13 Jul '11

13 Jul '11

> My knowledge of audio/alsa programming is really poor, but if someone suggest > a patch I can setup the thing and do some debug. Try Takashi 's daily snapshot tarball you should able to get "Headphone Playback Volume" after this patch "ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers" using hda-emu and it seem the driver support 6 channels even when the pink and blue jacks are not retasked

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 13 Jul '11

13 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 3 files changed, 11 insertions(+), 11 deletions(-)

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[alsa-devel] [PATCH 0/2] ASoC: STA32x: filter coefficients, bug fix
by Johannes Stezenbach 13 Jul '11

13 Jul '11

Hi, here are two patches for the ST STA32x ASoC codec driver. The first one adds SNDRV_CTL_ELEM_TYPE_BYTES controls for access to the biquad filter coefficients. I haven't found any other driver using SNDRV_CTL_ELEM_TYPE_BYTES in this way, but it works nicely. The coefficients can be set with amixer (although amixer has a quirk in that it prints the values in hex but can read them only as decimal). # amixer cget name='Ch1 - Biquad 1' numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00 # amixer cset name='Ch1 - Biquad 1' 12,34,0,0,0,0,0,0,0,0,0,0,0,255,78 numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x0c,0x22,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0xff,0x4e Due to the way the internal coeffcient RAM is accessed, the corresponding I2C registers are volatile, thus I added the .volatile_register function in this patch. The second patch preserves reset values as required by the chip documentation, it is mainly for future compatibility and was added during hardware testing to rule out issues from this side, although it did not make a difference in my tests. However, the chip spec insists it is important. Thanks Johannes

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