Author | Tokens | Token Proportion | Commits | Commit Proportion |
---|---|---|---|---|
Philippe Rétornaz | 3035 | 82.50% | 2 | 9.09% |
Steffen Trumtrar | 371 | 10.08% | 2 | 9.09% |
Alexander Shiyan | 97 | 2.64% | 4 | 18.18% |
Kuninori Morimoto | 81 | 2.20% | 2 | 9.09% |
Mark Brown | 34 | 0.92% | 2 | 9.09% |
Gaëtan Carlier | 30 | 0.82% | 1 | 4.55% |
Lars-Peter Clausen | 13 | 0.35% | 4 | 18.18% |
Takashi Iwai | 8 | 0.22% | 1 | 4.55% |
Axel Lin | 5 | 0.14% | 2 | 9.09% |
Fabio Estevam | 4 | 0.11% | 1 | 4.55% |
Bhumika Goyal | 1 | 0.03% | 1 | 4.55% |
Total | 3679 | 22 |
/* * 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 <linux/module.h> #include <linux/device.h> #include <linux/of.h> #include <linux/mfd/mc13xxx.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 <linux/regmap.h> #include "mc13783.h" #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 mc13xxx *mc13xxx; struct regmap *regmap; enum mc13783_ssi_port adc_ssi_port; enum mc13783_ssi_port dac_ssi_port; }; /* 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_component *component = dai->component; unsigned int rate = params_rate(params); int i; for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { if (rate == mc13783_rates[i]) { snd_soc_component_update_bits(component, 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_component *component = dai->component; 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_component_update_bits(component, 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_component *component = dai->component; 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_component_update_bits(component, 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_component *component = dai->component; 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_component_update_bits(component, 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_component *component = dai->component; 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 0x03: val |= SSI_NETWORK_DAC_RXSLOT_0_1; break; case 0x0c: val |= SSI_NETWORK_DAC_RXSLOT_2_3; break; case 0x30: val |= SSI_NETWORK_DAC_RXSLOT_4_5; break; case 0xc0: val |= SSI_NETWORK_DAC_RXSLOT_6_7; break; default: return -EINVAL; } snd_soc_component_update_bits(component, 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_component *component = dai->component; unsigned int val = 0; unsigned int mask = 0x3f; if (slots != 4) return -EINVAL; if (tx_mask != 0x3) return -EINVAL; val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ val |= (0x01 << 4); /* secondary timeslot TX is 1 */ snd_soc_component_update_bits(component, 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", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 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 SOC_ENUM_SINGLE_VIRT_DECL(adcl_enum, adcl_enum_text); static const struct snd_kcontrol_new left_input_mux = SOC_DAPM_ENUM("Route", adcl_enum); static const char * const adcr_enum_text[] = { "MC1R", "MC2", "RXINR", "TXIN", }; static SOC_ENUM_SINGLE_VIRT_DECL(adcr_enum, adcr_enum_text); static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const char * const speaker_amp_source_text[] = { "CODEC", "Right" }; static SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, speaker_amp_source_text); static const struct snd_kcontrol_new speaker_amp_source_mux = SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); static const char * const headset_amp_source_text[] = { "CODEC", "Mixer" }; static SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, headset_amp_source_text); static const struct snd_kcontrol_new headset_amp_source_mux = SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); static const struct snd_kcontrol_new cdcout_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0); static const struct snd_kcontrol_new adc_bypass_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_CODEC, 16, 1, 0); static const struct snd_kcontrol_new lamp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 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", MC13783_AUDIO_TX, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), SND_SOC_DAPM_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, &right_input_mux), SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0, &speaker_amp_source_mux), SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0, &headset_amp_source_mux), SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), SND_SOC_DAPM_PGA("Voice CODEC PGA", MC13783_AUDIO_RX1, 0, 0, NULL, 0), SND_SOC_DAPM_SWITCH("Voice CODEC Bypass", MC13783_AUDIO_CODEC, 16, 0, &adc_bypass_ctl), /* Output */ SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), SND_SOC_DAPM_OUTPUT("HSR"), SND_SOC_DAPM_OUTPUT("LSPL"), SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), SND_SOC_DAPM_OUTPUT("CDCOUT"), SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, &cdcout_ctl), SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, &hlamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, &hramp_ctl), SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, &llamp_ctl), SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, &lramp_ctl), SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 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"}, { "Voice CODEC PGA", "Voice CODEC Bypass", "ADC" }, { "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, { "Speaker Amp Source MUX", "Right", "DAC PGA"}, { "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"}, { "Headset Amp Source MUX", "Mixer", "DAC PGA"}, /* Output */ { "HSL", NULL, "Headset Amp Left" }, { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, { "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, { "Line out Amp Left", NULL, "DAC PGA"}, { "Line out Amp Right", NULL, "DAC PGA"}, { "DAC PGA", NULL, "DAC"}, { "DAC", NULL, "DAC_E"}, { "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"}, }; static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", "Mono", "Mono Mix"}; static SOC_ENUM_SINGLE_DECL(mc13783_enum_3d_mixer, MC13783_AUDIO_RX1, 16, 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_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_component *component) { struct mc13783_priv *priv = snd_soc_component_get_drvdata(component); snd_soc_component_init_regmap(component, dev_get_regmap(component->dev->parent, NULL)); /* 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); 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, AUDIO_SSI_SEL, 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, AUDIO_SSI_SEL, AUDIO_SSI_SEL); return 0; } static void mc13783_remove(struct snd_soc_component *component) { struct mc13783_priv *priv = snd_soc_component_get_drvdata(component); /* Make sure VAUDIOON is off */ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 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 const 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 const 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 = 2, .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 = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, }, .ops = &mc13783_ops_codec, }, }; static const 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 = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, }, .ops = &mc13783_ops_sync, .symmetric_rates = 1, } }; static const struct snd_soc_component_driver soc_component_dev_mc13783 = { .probe = mc13783_probe, .remove = mc13783_remove, .controls = mc13783_control_list, .num_controls = ARRAY_SIZE(mc13783_control_list), .dapm_widgets = mc13783_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(mc13783_dapm_widgets), .dapm_routes = mc13783_routes, .num_dapm_routes = ARRAY_SIZE(mc13783_routes), .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, }; static int __init mc13783_codec_probe(struct platform_device *pdev) { struct mc13783_priv *priv; struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; struct device_node *np; int ret; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; if (pdata) { priv->adc_ssi_port = pdata->adc_ssi_port; priv->dac_ssi_port = pdata->dac_ssi_port; } else { np = of_get_child_by_name(pdev->dev.parent->of_node, "codec"); if (!np) return -ENOSYS; ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) { of_node_put(np); return ret; } ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) { of_node_put(np); return ret; } of_node_put(np); } dev_set_drvdata(&pdev->dev, priv); priv->mc13xxx = dev_get_drvdata(pdev->dev.parent); if (priv->adc_ssi_port == priv->dac_ssi_port) ret = devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_mc13783, mc13783_dai_sync, ARRAY_SIZE(mc13783_dai_sync)); else ret = devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); return ret; } static int mc13783_codec_remove(struct platform_device *pdev) { return 0; } static struct platform_driver mc13783_codec_driver = { .driver = { .name = "mc13783-codec", }, .remove = mc13783_codec_remove, }; module_platform_driver_probe(mc13783_codec_driver, mc13783_codec_probe); MODULE_DESCRIPTION("ASoC MC13783 driver"); MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>"); MODULE_AUTHOR("Philippe Retornaz <philippe.retornaz@epfl.ch>"); MODULE_LICENSE("GPL");
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