From 8b908b8660f919a1a5135bc46acae12445767903 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Philippe=20R=C3=A9tornaz?= Date: Tue, 15 May 2012 13:53:50 +0200 Subject: [PATCH] ASoC: Add mc13783 codec MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Signed-off-by: Philippe Rétornaz Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 800 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 28 ++ 4 files changed, 834 insertions(+) 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 f8035bd09af6..22c686444633 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C @@ -444,6 +445,9 @@ config SND_SOC_MAX9768 config SND_SOC_MAX9877 tristate +config SND_SOC_MC13783 + tristate + config SND_SOC_ML26124 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d11ab901b075..a9663e9c375b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -31,6 +31,7 @@ snd-soc-max9768-objs := max9768.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-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o @@ -140,6 +141,7 @@ obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(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_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 000000000000..50fa38b9d183 --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,800 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer@pengutronix.de + * Copyright 2012 Philippe Retornaz, philippe.retornaz@epfl.ch + * + * 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 +#include +#include +#include +#include +#include +#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) +#define AUDIO_RX0_ADDSTDC (1 << 22) +#define AUDIO_RX0_ADDRXIN (1 << 23) + +#define AUDIO_RX1_PGARXEN (1 << 0); +#define AUDIO_RX1_PGASTEN (1 << 5) +#define AUDIO_RX1_ARXINEN (1 << 10) + +#define AUDIO_TX_AMC1REN (1 << 5) +#define AUDIO_TX_AMC1LEN (1 << 7) +#define AUDIO_TX_AMC2EN (1 << 9) +#define AUDIO_TX_ATXINEN (1 << 11) +#define AUDIO_TX_RXINREC (1 << 13) + +#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_DAC_SLOTS_8 (1 << 12) +#define SSI_NETWORK_DAC_SLOTS_4 (2 << 12) +#define SSI_NETWORK_DAC_SLOTS_2 (3 << 12) +#define SSI_NETWORK_DAC_SLOT_MASK (3 << 12) +#define SSI_NETWORK_DAC_RXSLOT_0_1 (0 << 14) +#define SSI_NETWORK_DAC_RXSLOT_2_3 (1 << 14) +#define SSI_NETWORK_DAC_RXSLOT_4_5 (2 << 14) +#define SSI_NETWORK_DAC_RXSLOT_6_7 (3 << 14) +#define SSI_NETWORK_DAC_RXSLOT_MASK (3 << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +/* + * MC13783_AUDIO_CODEC and MC13783_AUDIO_DAC mostly share the same + * register layout + */ +#define AUDIO_SSI_SEL (1 << 0) +#define AUDIO_CLK_SEL (1 << 1) +#define AUDIO_CSM (1 << 2) +#define AUDIO_BCL_INV (1 << 3) +#define AUDIO_CFS_INV (1 << 4) +#define AUDIO_CFS(x) (((x) & 0x3) << 5) +#define AUDIO_CLK(x) (((x) & 0x7) << 7) +#define AUDIO_C_EN (1 << 11) +#define AUDIO_C_CLK_EN (1 << 12) +#define AUDIO_C_RESET (1 << 15) + +#define AUDIO_CODEC_CDCFS8K16K (1 << 10) +#define AUDIO_DAC_CFS_DLY_B (1 << 10) + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13xxx *mc13xxx; + + 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, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params_dac(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; + + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, MC13783_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + + return -EINVAL; +} + +static int mc13783_pcm_hw_params_codec(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); + unsigned int val; + + switch (rate) { + case 8000: + val = 0; + break; + case 16000: + val = AUDIO_CODEC_CDCFS8K16K; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, MC13783_AUDIO_CODEC, AUDIO_CODEC_CDCFS8K16K, + val); + + return 0; +} + +static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return mc13783_pcm_hw_params_dac(substream, params, dai); + else + return mc13783_pcm_hw_params_codec(substream, params, dai); +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + unsigned int mask = AUDIO_CFS(3) | AUDIO_BCL_INV | AUDIO_CFS_INV | + AUDIO_CSM | AUDIO_C_CLK_EN | AUDIO_C_RESET; + + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= AUDIO_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + val |= AUDIO_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + val |= AUDIO_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + val |= AUDIO_BCL_INV | AUDIO_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + val |= AUDIO_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + val |= AUDIO_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + val |= AUDIO_CSM; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + val |= AUDIO_C_RESET; + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_fmt_async(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->id == MC13783_ID_STEREO_DAC) + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + else + return mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt) +{ + int ret; + + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_DAC); + if (ret) + return ret; + + /* + * In synchronous mode force the voice codec into slave mode + * so that the clock / framesync from the stereo DAC is used + */ + fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + fmt |= SND_SOC_DAIFMT_CBS_CFS; + ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); + + return ret; +} + +static int mc13783_sysclk[] = { + 13000000, + 15360000, + 16800000, + -1, + 26000000, + -1, /* 12000000, invalid for voice codec */ + -1, /* 3686400, invalid for voice codec */ + 33600000, +}; + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir, + unsigned int reg) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int val = 0; + unsigned int mask = AUDIO_CLK(0x7) | AUDIO_CLK_SEL; + + for (clk = 0; clk < ARRAY_SIZE(mc13783_sysclk); clk++) { + if (mc13783_sysclk[clk] < 0) + continue; + if (mc13783_sysclk[clk] == freq) + break; + } + + if (clk == ARRAY_SIZE(mc13783_sysclk)) + return -EINVAL; + + if (clk_id == MC13783_CLK_CLIB) + val |= AUDIO_CLK_SEL; + + val |= AUDIO_CLK(clk); + + snd_soc_update_bits(codec, reg, mask, val); + + return 0; +} + +static int mc13783_set_sysclk_dac(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); +} + +static int mc13783_set_sysclk_codec(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_sysclk_sync(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + int ret; + + ret = mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_DAC); + if (ret) + return ret; + + return mc13783_set_sysclk(dai, clk_id, freq, dir, MC13783_AUDIO_CODEC); +} + +static int mc13783_set_tdm_slot_dac(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 val = 0; + unsigned int mask = SSI_NETWORK_DAC_SLOT_MASK | + SSI_NETWORK_DAC_RXSLOT_MASK; + + switch (slots) { + case 2: + val |= SSI_NETWORK_DAC_SLOTS_2; + break; + case 4: + val |= SSI_NETWORK_DAC_SLOTS_4; + break; + case 8: + val |= SSI_NETWORK_DAC_SLOTS_8; + break; + default: + return -EINVAL; + } + + switch (rx_mask) { + case 0xfffffffc: + val |= SSI_NETWORK_DAC_RXSLOT_0_1; + break; + case 0xfffffff3: + val |= SSI_NETWORK_DAC_RXSLOT_2_3; + break; + case 0xffffffcf: + val |= SSI_NETWORK_DAC_RXSLOT_4_5; + break; + case 0xffffff3f: + val |= SSI_NETWORK_DAC_RXSLOT_6_7; + break; + default: + return -EINVAL; + }; + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + 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 val = 0; + unsigned int mask = 0x3f; + + if (slots != 4) + return -EINVAL; + + if (tx_mask != 0xfffffffc) + return -EINVAL; + + val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + val |= (0x01 << 4); /* secondary timeslot TX is 1 */ + + snd_soc_update_bits(codec, MC13783_SSI_NETWORK, mask, val); + + return 0; +} + +static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + int ret; + + ret = mc13783_set_tdm_slot_dac(dai, tx_mask, rx_mask, slots, + slot_width); + if (ret) + return ret; + + ret = mc13783_set_tdm_slot_codec(dai, tx_mask, rx_mask, slots, + slot_width); + + return ret; +} + +static const struct snd_kcontrol_new mc1l_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + +static const struct snd_kcontrol_new mc1r_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + +static const struct snd_kcontrol_new mc2_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + +static const struct snd_kcontrol_new atx_amp_ctl = + SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + + +/* Virtual mux. The chip does the input selection automatically + * as soon as we enable one input. */ +static const char * const adcl_enum_text[] = { + "MC1L", "RXINL", +}; + +static const struct soc_enum adcl_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text); + +static const struct snd_kcontrol_new left_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcl_enum); + +static const char * const adcr_enum_text[] = { + "MC1R", "MC2", "RXINR", "TXIN", +}; + +static const struct soc_enum adcr_enum = + SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text); + +static const struct snd_kcontrol_new right_input_mux = + SOC_DAPM_ENUM_VIRT("Route", adcr_enum); + +static const struct snd_kcontrol_new samp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + +static const struct snd_kcontrol_new lamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + +static const struct snd_kcontrol_new hlamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + +static const struct snd_kcontrol_new hramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + +static const struct snd_kcontrol_new llamp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + +static const struct snd_kcontrol_new lramp_ctl = + SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + +static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { +/* Input */ + SND_SOC_DAPM_INPUT("MC1LIN"), + SND_SOC_DAPM_INPUT("MC1RIN"), + SND_SOC_DAPM_INPUT("MC2IN"), + SND_SOC_DAPM_INPUT("RXINR"), + SND_SOC_DAPM_INPUT("RXINL"), + SND_SOC_DAPM_INPUT("TXIN"), + + SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + + SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, + &left_input_mux), + SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, + &right_input_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", 40, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + +/* Output */ + SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("RXOUTL"), + SND_SOC_DAPM_OUTPUT("RXOUTR"), + SND_SOC_DAPM_OUTPUT("HSL"), + SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSP"), + SND_SOC_DAPM_OUTPUT("SP"), + + SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), +}; + +static struct snd_soc_dapm_route mc13783_routes[] = { +/* Input */ + { "MC1L Amp", NULL, "MC1LIN"}, + { "MC1R Amp", NULL, "MC1RIN" }, + { "MC2 Amp", NULL, "MC2IN" }, + { "TXIN Amp", NULL, "TXIN"}, + + { "PGA Left Input Mux", "MC1L", "MC1L Amp" }, + { "PGA Left Input Mux", "RXINL", "RXINL"}, + { "PGA Right Input Mux", "MC1R", "MC1R Amp" }, + { "PGA Right Input Mux", "MC2", "MC2 Amp"}, + { "PGA Right Input Mux", "TXIN", "TXIN Amp"}, + { "PGA Right Input Mux", "RXINR", "RXINR"}, + + { "PGA Left Input", NULL, "PGA Left Input Mux"}, + { "PGA Right Input", NULL, "PGA Right Input Mux"}, + + { "ADC", NULL, "PGA Left Input"}, + { "ADC", NULL, "PGA Right Input"}, + { "ADC", NULL, "ADC_Reset"}, + +/* 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"}, + { "Line out Amp Left", NULL, "DAC PGA"}, + { "Line out Amp Right", NULL, "DAC PGA"}, + { "DAC PGA", NULL, "DAC"}, + { "DAC", NULL, "DAC_E"}, +}; + +static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_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_DOUBLE("PCM Capture Volume", MC13783_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); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + mc13xxx_lock(priv->mc13xxx); + + /* these are the reset values */ + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX1, 0x00d35A); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_TX, 0x420000); + mc13xxx_reg_write(priv->mc13xxx, MC13783_SSI_NETWORK, 0x013060); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_CODEC, 0x180027); + mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_DAC, 0x0e0004); + + snd_soc_add_codec_controls(codec, mc13783_control_list, + ARRAY_SIZE(mc13783_control_list)); + + snd_soc_dapm_new_controls(dapm, mc13783_dapm_widgets, + ARRAY_SIZE(mc13783_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, mc13783_routes, + ARRAY_SIZE(mc13783_routes)); + + if (priv->adc_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_CODEC, + 0, AUDIO_SSI_SEL); + + if (priv->dac_ssi_port == MC13783_SSI1_PORT) + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + AUDIO_SSI_SEL, 0); + else + mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, + 0, AUDIO_SSI_SEL); + + mc13xxx_unlock(priv->mc13xxx); + + return 0; +} + +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; +} + +#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) + +static struct snd_soc_dai_ops mc13783_ops_dac = { + .hw_params = mc13783_pcm_hw_params_dac, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_dac, + .set_tdm_slot = mc13783_set_tdm_slot_dac, +}; + +static struct snd_soc_dai_ops mc13783_ops_codec = { + .hw_params = mc13783_pcm_hw_params_codec, + .set_fmt = mc13783_set_fmt_async, + .set_sysclk = mc13783_set_sysclk_codec, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +/* + * The mc13783 has two SSI ports, both of them can be routed either + * to the voice codec or the stereo DAC. When two different SSI ports + * are used for the voice codec and the stereo DAC we can do different + * formats and sysclock settings for playback and capture + * (mc13783-hifi-playback and mc13783-hifi-capture). Using the same port + * forces us to use symmetric rates (mc13783-hifi). + */ +static struct snd_soc_dai_driver mc13783_dai_async[] = { + { + .name = "mc13783-hifi-playback", + .id = MC13783_ID_STEREO_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_dac, + }, { + .name = "mc13783-hifi-capture", + .id = MC13783_ID_STEREO_CODEC, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops_codec, + }, +}; + +static struct snd_soc_dai_ops mc13783_ops_sync = { + .hw_params = mc13783_pcm_hw_params_sync, + .set_fmt = mc13783_set_fmt_sync, + .set_sysclk = mc13783_set_sysclk_sync, + .set_tdm_slot = mc13783_set_tdm_slot_sync, +}; + +static struct snd_soc_dai_driver mc13783_dai_sync[] = { + { + .name = "mc13783-hifi", + .id = MC13783_ID_SYNC, + .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_sync, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, + .read = mc13783_read, + .write = mc13783_write, +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13xxx *mc13xxx; + struct mc13783_priv *priv; + struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; + int ret; + + mc13xxx = dev_get_drvdata(pdev->dev.parent); + + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13xxx = mc13xxx; + if (pdata) { + priv->adc_ssi_port = pdata->adc_ssi_port; + priv->dac_ssi_port = pdata->dac_ssi_port; + } else { + priv->adc_ssi_port = MC13783_SSI1_PORT; + priv->dac_ssi_port = MC13783_SSI2_PORT; + } + + if (priv->adc_ssi_port == priv->dac_ssi_port) + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync)); + else + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, + mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); + + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + + 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 "); +MODULE_AUTHOR("Philippe Retornaz "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 000000000000..3a6d1993a217 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,28 @@ +/* + * Copyright 2008 Juergen Beisert, kernel@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 + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 +#define MC13783_ID_SYNC 3 + +#endif /* MC13783_MIXER_H */ -- 2.20.1