/* * uda1380.c - Philips UDA1380 ALSA SoC audio driver * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * * Copyright (c) 2007 Philipp Zabel * Improved support for DAPM and audio routing/mixing capabilities, * added TLV support. * * Modified by Richard Purdie to fit into SoC * codec model. * * Copyright (c) 2005 Giorgio Padrin * Copyright 2005 Openedhand Ltd. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "uda1380.h" #define UDA1380_VERSION "0.6" /* * uda1380 register cache */ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { 0x0502, 0x0000, 0x0000, 0x3f3f, 0x0202, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0xff00, 0x0000, 0x4800, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x8000, 0x0002, 0x0000, }; /* * read uda1380 register cache */ static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; if (reg == UDA1380_RESET) return 0; if (reg >= UDA1380_CACHEREGNUM) return -1; return cache[reg]; } /* * write uda1380 register cache */ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; if (reg >= UDA1380_CACHEREGNUM) return; cache[reg] = value; } /* * write to the UDA1380 register space */ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[3]; /* data is * data[0] is register offset * data[1] is MS byte * data[2] is LS byte */ data[0] = reg; data[1] = (value & 0xff00) >> 8; data[2] = value & 0x00ff; uda1380_write_reg_cache(codec, reg, value); /* the interpolator & decimator regs must only be written when the * codec DAI is active. */ if (!codec->active && (reg >= UDA1380_MVOL)) return 0; pr_debug("uda1380: hw write %x val %x\n", reg, value); if (codec->hw_write(codec->control_data, data, 3) == 3) { unsigned int val; i2c_master_send(codec->control_data, data, 1); i2c_master_recv(codec->control_data, data, 2); val = (data[0]<<8) | data[1]; if (val != value) { pr_debug("uda1380: READ BACK VAL %x\n", (data[0]<<8) | data[1]); return -EIO; } return 0; } else return -EIO; } #define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) /* declarations of ALSA reg_elem_REAL controls */ static const char *uda1380_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz", "96kHz", }; static const char *uda1380_input_sel[] = { "Line", "Mic + Line R", "Line L", "Mic", }; static const char *uda1380_output_sel[] = { "DAC", "Analog Mixer", }; static const char *uda1380_spf_mode[] = { "Flat", "Minimum1", "Minimum2", "Maximum" }; static const char *uda1380_capture_sel[] = { "ADC", "Digital Mixer" }; static const char *uda1380_sel_ns[] = { "3rd-order", "5th-order" }; static const char *uda1380_mix_control[] = { "off", "PCM only", "before sound processing", "after sound processing" }; static const char *uda1380_sdet_setting[] = { "3200", "4800", "9600", "19200" }; static const char *uda1380_os_setting[] = { "single-speed", "double-speed (no mixing)", "quad-speed (no mixing)" }; static const struct soc_enum uda1380_deemp_enum[] = { SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), }; static const struct soc_enum uda1380_input_sel_enum = SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ static const struct soc_enum uda1380_output_sel_enum = SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ static const struct soc_enum uda1380_spf_enum = SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ static const struct soc_enum uda1380_capture_sel_enum = SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ static const struct soc_enum uda1380_sel_ns_enum = SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ static const struct soc_enum uda1380_mix_enum = SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ static const struct soc_enum uda1380_sdet_enum = SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ static const struct soc_enum uda1380_os_enum = SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ /* * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) */ static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); /* * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored), * from -66 dB in 0.5 dB steps (2 dB steps, really) and * from -52 dB in 0.25 dB steps */ static const unsigned int mvol_tlv[] = { TLV_DB_RANGE_HEAD(3), 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1), 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0), 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0), }; /* * from -72 dB in 1.5 dB steps (6 dB steps really), * from -66 dB in 0.75 dB steps (3 dB steps really), * from -60 dB in 0.5 dB steps (2 dB steps really) and * from -46 dB in 0.25 dB steps */ static const unsigned int vc_tlv[] = { TLV_DB_RANGE_HEAD(4), 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1), 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0), 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0), 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0), }; /* from 0 to 6 dB in 2 dB steps if SPF mode != flat */ static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); /* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts * off at 18 dB max) */ static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); /* from -63 to 24 dB in 0.5 dB steps (-128...48) */ static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1); /* from 0 to 24 dB in 3 dB steps */ static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); /* from 0 to 30 dB in 2 dB steps */ static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */ SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */ SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */ SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */ SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */ SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */ SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */ /**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */ SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */ SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */ SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */ SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */ SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */ /**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */ SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */ SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */ SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */ SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */ SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */ /* -5.5, -8, -11.5, -14 dBFS */ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), }; /* add non dapm controls */ static int uda1380_add_controls(struct snd_soc_codec *codec) { int err, i; for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { err = snd_ctl_add(codec->card, snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); if (err < 0) return err; } return 0; } /* Input mux */ static const struct snd_kcontrol_new uda1380_input_mux_control = SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); /* Output mux */ static const struct snd_kcontrol_new uda1380_output_mux_control = SOC_DAPM_ENUM("Route", uda1380_output_sel_enum); /* Capture mux */ static const struct snd_kcontrol_new uda1380_capture_mux_control = SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum); static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &uda1380_input_mux_control), SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0, &uda1380_output_mux_control), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &uda1380_capture_mux_control), SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), SND_SOC_DAPM_INPUT("VINM"), SND_SOC_DAPM_INPUT("VINL"), SND_SOC_DAPM_INPUT("VINR"), SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("VOUTLHP"), SND_SOC_DAPM_OUTPUT("VOUTRHP"), SND_SOC_DAPM_OUTPUT("VOUTL"), SND_SOC_DAPM_OUTPUT("VOUTR"), SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), }; static const struct snd_soc_dapm_route audio_map[] = { /* output mux */ {"HeadPhone Driver", NULL, "Output Mux"}, {"VOUTR", NULL, "Output Mux"}, {"VOUTL", NULL, "Output Mux"}, {"Analog Mixer", NULL, "VINR"}, {"Analog Mixer", NULL, "VINL"}, {"Analog Mixer", NULL, "DAC"}, {"Output Mux", "DAC", "DAC"}, {"Output Mux", "Analog Mixer", "Analog Mixer"}, /* {"DAC", "Digital Mixer", "I2S" } */ /* headphone driver */ {"VOUTLHP", NULL, "HeadPhone Driver"}, {"VOUTRHP", NULL, "HeadPhone Driver"}, /* input mux */ {"Left ADC", NULL, "Input Mux"}, {"Input Mux", "Mic", "Mic LNA"}, {"Input Mux", "Mic + Line R", "Mic LNA"}, {"Input Mux", "Line L", "Left PGA"}, {"Input Mux", "Line", "Left PGA"}, /* right input */ {"Right ADC", "Mic + Line R", "Right PGA"}, {"Right ADC", "Line", "Right PGA"}, /* inputs */ {"Mic LNA", NULL, "VINM"}, {"Left PGA", NULL, "VINL"}, {"Right PGA", NULL, "VINR"}, }; static int uda1380_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, ARRAY_SIZE(uda1380_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; } static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; int iface; /* set up DAI based upon fmt */ iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface |= R01_SFORI_I2S | R01_SFORO_I2S; break; case SND_SOC_DAIFMT_LSB: iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; break; case SND_SOC_DAIFMT_MSB: iface |= R01_SFORI_MSB | R01_SFORO_I2S; } if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) iface |= R01_SIM; uda1380_write(codec, UDA1380_IFACE, iface); return 0; } /* * Flush reg cache * We can only write the interpolator and decimator registers * when the DAI is being clocked by the CPU DAI. It's up to the * machine and cpu DAI driver to do this before we are called. */ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; int reg, reg_start, reg_end, clk; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg_start = UDA1380_MVOL; reg_end = UDA1380_MIXER; } else { reg_start = UDA1380_DEC; reg_end = UDA1380_AGC; } /* FIXME disable DAC_CLK */ clk = uda1380_read_reg_cache(codec, UDA1380_CLK); uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); for (reg = reg_start; reg <= reg_end; reg++) { pr_debug("uda1380: flush reg %x val %x:", reg, uda1380_read_reg_cache(codec, reg)); uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); } /* FIXME enable DAC_CLK */ uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); return 0; } static int uda1380_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_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ if (clk & R00_DAC_CLK) { int rate = params_rate(params); u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); clk &= ~0x3; /* clear SEL_LOOP_DIV */ switch (rate) { case 6250 ... 12500: clk |= 0x0; break; case 12501 ... 25000: clk |= 0x1; break; case 25001 ... 50000: clk |= 0x2; break; case 50001 ... 100000: clk |= 0x3; break; } uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clk |= R00_EN_DAC | R00_EN_INT; else clk |= R00_EN_ADC | R00_EN_DEC; uda1380_write(codec, UDA1380_CLK, clk); return 0; } static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ if (clk & R00_DAC_CLK) { u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clk &= ~(R00_EN_DAC | R00_EN_INT); else clk &= ~(R00_EN_ADC | R00_EN_DEC); uda1380_write(codec, UDA1380_CLK, clk); } static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; /* FIXME: mute(codec,0) is called when the magician clock is already * set to WSPLL, but for some unknown reason writing to interpolator * registers works only when clocked by SYSCLK */ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); if (mute) uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); else uda1380_write(codec, UDA1380_DEEMP, mute_reg); uda1380_write(codec, UDA1380_CLK, clk); return 0; } static int uda1380_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { int pm = uda1380_read_reg_cache(codec, UDA1380_PM); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); break; case SND_SOC_BIAS_STANDBY: uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); break; case SND_SOC_BIAS_OFF: uda1380_write(codec, UDA1380_PM, 0x0); break; } codec->bias_level = level; return 0; } #define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, }, { /* playback only - dual interface */ .name = "UDA1380", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, }, { /* capture only - dual interface*/ .name = "UDA1380", .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, .set_fmt = uda1380_set_dai_fmt, }, }, }; EXPORT_SYMBOL_GPL(uda1380_dai); static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int uda1380_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); } uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda1380_set_bias_level(codec, codec->suspend_bias_level); return 0; } /* * initialise the UDA1380 driver * register mixer and dsp interfaces with the kernel */ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) { struct snd_soc_codec *codec = socdev->codec; int ret = 0; codec->name = "UDA1380"; codec->owner = THIS_MODULE; codec->read = uda1380_read_reg_cache; codec->write = uda1380_write; codec->set_bias_level = uda1380_set_bias_level; codec->dai = uda1380_dai; codec->num_dai = ARRAY_SIZE(uda1380_dai); codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); codec->reg_cache_step = 1; uda1380_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { pr_err("uda1380: failed to create pcms\n"); goto pcm_err; } /* power on device */ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set clock input */ switch (dac_clk) { case UDA1380_DAC_CLK_SYSCLK: uda1380_write(codec, UDA1380_CLK, 0); break; case UDA1380_DAC_CLK_WSPLL: uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK); break; } /* uda1380 init */ uda1380_add_controls(codec); uda1380_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { pr_err("uda1380: failed to register card\n"); goto card_err; } return ret; card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; } static struct snd_soc_device *uda1380_socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int uda1380_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = uda1380_socdev; struct uda1380_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; int ret; i2c_set_clientdata(i2c, codec); codec->control_data = i2c; ret = uda1380_init(socdev, setup->dac_clk); if (ret < 0) pr_err("uda1380: failed to initialise UDA1380\n"); return ret; } static int uda1380_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); kfree(codec->reg_cache); return 0; } static const struct i2c_device_id uda1380_i2c_id[] = { { "uda1380", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); static struct i2c_driver uda1380_i2c_driver = { .driver = { .name = "UDA1380 I2C Codec", .owner = THIS_MODULE, }, .probe = uda1380_i2c_probe, .remove = uda1380_i2c_remove, .id_table = uda1380_i2c_id, }; static int uda1380_add_i2c_device(struct platform_device *pdev, const struct uda1380_setup_data *setup) { struct i2c_board_info info; struct i2c_adapter *adapter; struct i2c_client *client; int ret; ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) { dev_err(&pdev->dev, "can't add i2c driver\n"); return ret; } memset(&info, 0, sizeof(struct i2c_board_info)); info.addr = setup->i2c_address; strlcpy(info.type, "uda1380", I2C_NAME_SIZE); adapter = i2c_get_adapter(setup->i2c_bus); if (!adapter) { dev_err(&pdev->dev, "can't get i2c adapter %d\n", setup->i2c_bus); goto err_driver; } client = i2c_new_device(adapter, &info); i2c_put_adapter(adapter); if (!client) { dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", (unsigned int)info.addr); goto err_driver; } return 0; err_driver: i2c_del_driver(&uda1380_i2c_driver); return -ENODEV; } #endif static int uda1380_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct uda1380_setup_data *setup; struct snd_soc_codec *codec; int ret; pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); uda1380_socdev = socdev; ret = -ENODEV; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { codec->hw_write = (hw_write_t)i2c_master_send; ret = uda1380_add_i2c_device(pdev, setup); } #endif if (ret != 0) kfree(codec); return ret; } /* power down chip */ static int uda1380_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_unregister_device(codec->control_data); i2c_del_driver(&uda1380_i2c_driver); #endif kfree(codec); return 0; } struct snd_soc_codec_device soc_codec_dev_uda1380 = { .probe = uda1380_probe, .remove = uda1380_remove, .suspend = uda1380_suspend, .resume = uda1380_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); static int __init uda1380_modinit(void) { return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); } module_init(uda1380_modinit); static void __exit uda1380_exit(void) { snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); } module_exit(uda1380_exit); MODULE_AUTHOR("Giorgio Padrin"); MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); MODULE_LICENSE("GPL");