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author | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-29 01:25:01 +0400 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2011-10-29 01:25:01 +0400 |
commit | 68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch) | |
tree | f189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound/soc/codecs/da7210.c | |
parent | 0e59e7e7feb5a12938fbf9135147eeda3238c6c4 (diff) | |
parent | 8128c9f21509f9a8b6da94ac432d845dda458406 (diff) | |
download | linux-68d99b2c8efcb6ed3807a55569300c53b5f88be5.tar.xz |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
Diffstat (limited to 'sound/soc/codecs/da7210.c')
-rw-r--r-- | sound/soc/codecs/da7210.c | 649 |
1 files changed, 515 insertions, 134 deletions
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 92fd9d7a9221..0ebcbd534490 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,23 +26,41 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -50,6 +68,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -72,6 +96,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -105,12 +130,17 @@ /* DAI_CFG1 bit fields */ #define DA7210_DAI_WORD_S16_LE (0 << 0) +#define DA7210_DAI_WORD_S20_3LE (1 << 0) #define DA7210_DAI_WORD_S24_LE (2 << 0) +#define DA7210_DAI_WORD_S32_LE (3 << 0) #define DA7210_DAI_FLEN_64BIT (1 << 2) +#define DA7210_DAI_MODE_SLAVE (0 << 7) #define DA7210_DAI_MODE_MASTER (1 << 7) /* DAI_CFG3 bit fields */ #define DA7210_DAI_FORMAT_I2SMODE (0 << 0) +#define DA7210_DAI_FORMAT_LEFT_J (1 << 0) +#define DA7210_DAI_FORMAT_RIGHT_J (2 << 0) #define DA7210_DAI_OE (1 << 3) #define DA7210_DAI_EN (1 << 7) @@ -133,6 +163,43 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -144,24 +211,351 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +/* ADC and DAC high pass filter f0 value */ +static const char const *da7210_hpf_cutoff_txt[] = { + "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" +}; + +static const struct soc_enum da7210_dac_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +static const struct soc_enum da7210_adc_hpf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); + +/* ADC and DAC voice (8kHz) high pass cutoff value */ +static const char const *da7210_vf_cutoff_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7210_dac_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const struct soc_enum da7210_adc_vf_cutoff = + SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); + +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EINVAL; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC | + DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EINVAL; +} static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0), + SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff), + SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0), + SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff), + + SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0), + SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), + SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), + SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), +}; + +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, }; /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* @@ -188,72 +582,15 @@ static const u8 da7210_reg[] = { 0x00, /* R88 */ }; -/* - * Read da7210 register cache - */ -static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) -{ - u8 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); - return cache[reg]; -} - -/* - * Write to the da7210 register space - */ -static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) +static int da7210_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { - u8 *cache = codec->reg_cache; - u8 data[2]; - - BUG_ON(codec->driver->volatile_register); - - data[0] = reg & 0xff; - data[1] = value & 0xff; - - if (reg >= codec->driver->reg_cache_size) - return -EIO; - - if (2 != codec->hw_write(codec->control_data, data, 2)) - return -EIO; - - cache[reg] = value; - return 0; -} - -/* - * Read from the da7210 register space. - */ -static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) -{ - if (DA7210_STATUS == reg) - return i2c_smbus_read_byte_data(codec->control_data, reg); - - return da7210_read_reg_cache(codec, reg); -} - -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + switch (reg) { + case DA7210_STATUS: + return 1; + default: + return 0; } - - return 0; } /* @@ -266,93 +603,75 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; u32 dai_cfg1; - u32 hpf_reg, hpf_mask, hpf_value; u32 fs, bypass; /* set DAI source to Left and Right ADC */ - da7210_write(codec, DA7210_DAI_SRC_SEL, + snd_soc_write(codec, DA7210_DAI_SRC_SEL, DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); /* Enable DAI */ - da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: dai_cfg1 |= DA7210_DAI_WORD_S16_LE; break; + case SNDRV_PCM_FORMAT_S20_3LE: + dai_cfg1 |= DA7210_DAI_WORD_S20_3LE; + break; case SNDRV_PCM_FORMAT_S24_LE: dai_cfg1 |= DA7210_DAI_WORD_S24_LE; break; + case SNDRV_PCM_FORMAT_S32_LE: + dai_cfg1 |= DA7210_DAI_WORD_S32_LE; + break; default: return -EINVAL; } - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - - hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? - DA7210_DAC_HPF : DA7210_ADC_HPF; + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; - hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - hpf_mask = DA7210_VOICE_EN; - hpf_value = 0; bypass = DA7210_PLL_BYP; break; default: @@ -362,7 +681,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, /* Disable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); - snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); @@ -382,13 +700,16 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) u32 dai_cfg1; u32 dai_cfg3; - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; + case SND_SOC_DAIFMT_CBS_CFS: + dai_cfg1 |= DA7210_DAI_MODE_SLAVE; + break; default: return -EINVAL; } @@ -401,6 +722,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) case SND_SOC_DAIFMT_I2S: dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE; break; + case SND_SOC_DAIFMT_LEFT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J; + break; default: return -EINVAL; } @@ -411,19 +738,32 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) */ dai_cfg1 |= DA7210_DAI_FLEN_64BIT; - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); + + return 0; +} + +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); return 0; } -#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -451,11 +791,15 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; - codec->hw_write = (hw_write_t)i2c_master_send; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* FIXME * @@ -472,8 +816,8 @@ static int da7210_probe(struct snd_soc_codec *codec) /* * make sure that DA7210 use bypass mode before start up */ - da7210_write(codec, DA7210_STARTUP1, 0); - da7210_write(codec, DA7210_PLL_DIV3, + snd_soc_write(codec, DA7210_STARTUP1, 0); + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* @@ -481,36 +825,70 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* Enable Left & Right MIC PGA and Mic Bias */ - da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); - da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); /* Enable Left and Right input PGA */ - da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); - da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); + snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); /* Enable Left and Right ADC */ - da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); + snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); /* * DAC settings */ /* Enable Left and Right DAC */ - da7210_write(codec, DA7210_DAC_SEL, + snd_soc_write(codec, DA7210_DAC_SEL, DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); /* Enable Left and Right out PGA */ - da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); - da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); /* Enable Left and Right HeadPhone PGA */ - da7210_write(codec, DA7210_HP_CFG, + snd_soc_write(codec, DA7210_HP_CFG, DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + + /* + * For DA7210 codec, there are two ways to enable/disable analog IOs + * and ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO + * (or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) + * + * Out of these two methods, the one using STANDBY bits is preferred + * way to enable/disable individual blocks. This is because STANDBY + * registers are part of system controller which allows system power + * up/down in a controlled, pop-free manner. Also, as per application + * note of DA7210, STANDBY register bits are only effective if a + * particular IO (or ADC/DAC) is already enabled using enable/disable + * register bits. Keeping these things in mind, current DAPM + * implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() + * - "STANDBY bit" is controlled by DAPM + */ + + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ - da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); /* * If 48kHz sound came, it use bypass mode, @@ -521,25 +899,22 @@ static int da7210_probe(struct snd_soc_codec *codec) * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. * see da7210_hw_params */ - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - da7210_write(codec, DA7210_PLL_DIV2, 0x99); - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - da7210_write(codec, DA7210_A_PLL1, 0x01); - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + snd_soc_write(codec, DA7210_A_PLL1, 0x01); + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); - - snd_soc_add_controls(codec, da7210_snd_controls, - ARRAY_SIZE(da7210_snd_controls)); + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); @@ -548,11 +923,18 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, .reg_cache_size = ARRAY_SIZE(da7210_reg), .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, + .volatile_register = da7210_volatile_register, + + .controls = da7210_snd_controls, + .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -567,7 +949,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_data = i2c; da7210->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, |