From 368494093354ac613a80c2e1d77602aa12473cf0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Nov 2014 12:27:33 +0200 Subject: ASoC: tlv320aic3x: Add TDM support TDM support is achieved using DSP transfer mode and setting a programmable offset which specifies where data begins with respect to the frame sync. It requires 256-clock mode if CODEC is master (not currently supported in the driver). No additional dependency if CODEC is slave. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f7c2a57..8770e28 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -78,6 +78,8 @@ struct aic3x_priv { struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; struct aic3x_setup_data *setup; unsigned int sysclk; + unsigned int dai_fmt; + unsigned int tdm_delay; struct list_head list; int master; int gpio_reset; @@ -1009,6 +1011,25 @@ found: return 0; } +static int aic3x_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + int delay = 0; + + /* TDM slot selection only valid in DSP_A/_B mode */ + if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A) + delay += (aic3x->tdm_delay + 1); + else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B) + delay += aic3x->tdm_delay; + + /* Configure data delay */ + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + + return 0; +} + static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -1048,7 +1069,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); u8 iface_areg, iface_breg; - int delay = 0; iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -1076,7 +1096,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): - delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -1090,10 +1109,45 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + aic3x->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* set iface */ snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); + + return 0; +} + +static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + unsigned int lsb; + + if (tx_mask != rx_mask) { + dev_err(codec->dev, "tx and rx masks must be symmetric\n"); + return -EINVAL; + } + + if (unlikely(!tx_mask)) { + dev_err(codec->dev, "tx and rx masks need to be non 0\n"); + return -EINVAL; + } + + /* TDM based on DSP mode requires slots to be adjacent */ + lsb = __ffs(tx_mask); + if ((lsb + 1) != __fls(tx_mask)) { + dev_err(codec->dev, "Invalid mask, slots must be adjacent\n"); + return -EINVAL; + } + + aic3x->tdm_delay = lsb * slot_width; + + /* DOUT in high-impedance on inactive bit clocks */ + snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA, + DOUT_TRISTATE, DOUT_TRISTATE); return 0; } @@ -1212,9 +1266,11 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops aic3x_dai_ops = { .hw_params = aic3x_hw_params, + .prepare = aic3x_prepare, .digital_mute = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, .set_fmt = aic3x_set_dai_fmt, + .set_tdm_slot = aic3x_set_dai_tdm_slot, }; static struct snd_soc_dai_driver aic3x_dai = { diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e521ac3..89fa692 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -169,6 +169,7 @@ /* Audio serial data interface control register A bits */ #define BIT_CLK_MASTER 0x80 #define WORD_CLK_MASTER 0x40 +#define DOUT_TRISTATE 0x20 /* Codec Datapath setup register 7 */ #define FSREF_44100 (1 << 7) -- cgit v0.10.2 From a60e654be733a69879148cb4c56d0f58b749e3c4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 11 Nov 2014 10:59:00 +0200 Subject: ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual It is easier to find the relevant enums in the code. Use the SOC_ENUM_*_DECL macro for the individual items. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8770e28..9901400 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -216,61 +216,68 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; -static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; -static const char *aic3x_left_hpcom_mux[] = - { "differential of HPLOUT", "constant VCM", "single-ended" }; -static const char *aic3x_right_hpcom_mux[] = - { "differential of HPROUT", "constant VCM", "single-ended", - "differential of HPLCOM", "external feedback" }; -static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; -static const char *aic3x_adc_hpf[] = - { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; - -#define LDAC_ENUM 0 -#define RDAC_ENUM 1 -#define LHPCOM_ENUM 2 -#define RHPCOM_ENUM 3 -#define LINE1L_2_L_ENUM 4 -#define LINE1L_2_R_ENUM 5 -#define LINE1R_2_L_ENUM 6 -#define LINE1R_2_R_ENUM 7 -#define LINE2L_ENUM 8 -#define LINE2R_ENUM 9 -#define ADC_HPF_ENUM 10 - -static const struct soc_enum aic3x_enum[] = { - SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), - SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux), - SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux), - SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux), - SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), -}; - -static const char *aic3x_agc_level[] = - { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; -static const struct soc_enum aic3x_agc_level_enum[] = { - SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), - SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), -}; - -static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; -static const struct soc_enum aic3x_agc_attack_enum[] = { - SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), - SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), -}; - -static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; -static const struct soc_enum aic3x_agc_decay_enum[] = { - SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), - SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), -}; +static const char * const aic3x_left_dac_mux[] = { + "DAC_L1", "DAC_L3", "DAC_L2" }; +static SOC_ENUM_SINGLE_DECL(aic3x_left_dac_enum, DAC_LINE_MUX, 6, + aic3x_left_dac_mux); + +static const char * const aic3x_right_dac_mux[] = { + "DAC_R1", "DAC_R3", "DAC_R2" }; +static SOC_ENUM_SINGLE_DECL(aic3x_right_dac_enum, DAC_LINE_MUX, 4, + aic3x_right_dac_mux); + +static const char * const aic3x_left_hpcom_mux[] = { + "differential of HPLOUT", "constant VCM", "single-ended" }; +static SOC_ENUM_SINGLE_DECL(aic3x_left_hpcom_enum, HPLCOM_CFG, 4, + aic3x_left_hpcom_mux); + +static const char * const aic3x_right_hpcom_mux[] = { + "differential of HPROUT", "constant VCM", "single-ended", + "differential of HPLCOM", "external feedback" }; +static SOC_ENUM_SINGLE_DECL(aic3x_right_hpcom_enum, HPRCOM_CFG, 3, + aic3x_right_hpcom_mux); + +static const char * const aic3x_linein_mode_mux[] = { + "single-ended", "differential" }; +static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_l_enum, LINE1L_2_LADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_r_enum, LINE1L_2_RADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_l_enum, LINE1R_2_LADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_r_enum, LINE1R_2_RADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line2l_2_ldac_enum, LINE2L_2_LADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line2r_2_rdac_enum, LINE2R_2_RADC_CTRL, 7, + aic3x_linein_mode_mux); + +static const char * const aic3x_adc_hpf[] = { + "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static SOC_ENUM_DOUBLE_DECL(aic3x_adc_hpf_enum, AIC3X_CODEC_DFILT_CTRL, 6, 4, + aic3x_adc_hpf); + +static const char * const aic3x_agc_level[] = { + "-5.5dB", "-8dB", "-10dB", "-12dB", + "-14dB", "-17dB", "-20dB", "-24dB" }; +static SOC_ENUM_SINGLE_DECL(aic3x_lagc_level_enum, LAGC_CTRL_A, 4, + aic3x_agc_level); +static SOC_ENUM_SINGLE_DECL(aic3x_ragc_level_enum, RAGC_CTRL_A, 4, + aic3x_agc_level); + +static const char * const aic3x_agc_attack[] = { + "8ms", "11ms", "16ms", "20ms" }; +static SOC_ENUM_SINGLE_DECL(aic3x_lagc_attack_enum, LAGC_CTRL_A, 2, + aic3x_agc_attack); +static SOC_ENUM_SINGLE_DECL(aic3x_ragc_attack_enum, RAGC_CTRL_A, 2, + aic3x_agc_attack); + +static const char * const aic3x_agc_decay[] = { + "100ms", "200ms", "400ms", "500ms" }; +static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0, + aic3x_agc_decay); +static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0, + aic3x_agc_decay); /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps @@ -385,12 +392,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), - SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), - SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), - SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), - SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), - SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), - SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), + SOC_ENUM("Left AGC Target level", aic3x_lagc_level_enum), + SOC_ENUM("Right AGC Target level", aic3x_ragc_level_enum), + SOC_ENUM("Left AGC Attack time", aic3x_lagc_attack_enum), + SOC_ENUM("Right AGC Attack time", aic3x_ragc_attack_enum), + SOC_ENUM("Left AGC Decay time", aic3x_lagc_decay_enum), + SOC_ENUM("Right AGC Decay time", aic3x_ragc_decay_enum), /* De-emphasis */ SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), @@ -400,7 +407,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { 0, 119, 0, adc_tlv), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), - SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { @@ -427,19 +434,19 @@ static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_left_dac_enum); /* Right DAC Mux */ static const struct snd_kcontrol_new aic3x_right_dac_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_right_dac_enum); /* Left HPCOM Mux */ static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_left_hpcom_enum); /* Right HPCOM Mux */ static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum); /* Left Line Mixer */ static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { @@ -531,23 +538,23 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { /* Left Line1 Mux */ static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum); static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1l_2_r_enum); /* Right Line1 Mux */ static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1r_2_r_enum); static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1r_2_l_enum); /* Left Line2 Mux */ static const struct snd_kcontrol_new aic3x_left_line2_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line2l_2_ldac_enum); /* Right Line2 Mux */ static const struct snd_kcontrol_new aic3x_right_line2_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line2r_2_rdac_enum); static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Left DAC to Left Outputs */ -- cgit v0.10.2 From 68d6626925c3529790a2055d41578415fa98495e Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Tue, 11 Nov 2014 10:59:01 +0200 Subject: ASoC: tlv320aic3x: Add output driver pop reduction controls Output driver has two parameters that can be configured to reduce pop noise: power-on delay and ramp-up step time. Two new kcontrols have been added to set these parameters. Signed-off-by: Misael Lopez Cruz Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 9901400..f0a8281 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -279,6 +279,16 @@ static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0, static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0, aic3x_agc_decay); +static const char * const aic3x_poweron_time[] = { + "0us", "10us", "100us", "1ms", "10ms", "50ms", + "100ms", "200ms", "400ms", "800ms", "2s", "4s" }; +static SOC_ENUM_SINGLE_DECL(aic3x_poweron_time_enum, HPOUT_POP_REDUCTION, 4, + aic3x_poweron_time); + +static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" }; +static SOC_ENUM_SINGLE_DECL(aic3x_rampup_step_enum, HPOUT_POP_REDUCTION, 2, + aic3x_rampup_step); + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -408,6 +418,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum), + + /* Pop reduction */ + SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), + SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { -- cgit v0.10.2 From 2880fc877971d6c14b0c76ac09744e3ff5b126d5 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 13 Nov 2014 11:18:29 -0800 Subject: ASoC: add TI ts3a227e headset chip driver The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on the jack as buttons 0-3 mapped to KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND. This will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..1d12057 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,314 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * 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. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + bool plugged; + bool mic_present; + unsigned int buttons_held; +}; + +/* Button values to be reported on the jack */ +static const int ts3a227e_buttons[] = { + SND_JACK_BTN_0, + SND_JACK_BTN_1, + SND_JACK_BTN_2, + SND_JACK_BTN_3, +}; + +#define TS3A227E_NUM_BUTTONS 4 +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_MICROPHONE | \ + SND_JACK_BTN_0 | \ + SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | \ + SND_JACK_BTN_3) + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + unsigned int i; + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (ts3a227e->buttons_held & (1 << i)) + report |= ts3a227e_buttons[i]; + } + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + ts3a227e->buttons_held = 0; + if (mic_present) { + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, + TS3A227E_REG_SETTING_2, + KP_ENABLE, KP_ENABLE); + } + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + ts3a227e->buttons_held |= (1 << i); + if (kp_int_reg & RELEASE_MASK(i)) + ts3a227e->buttons_held &= ~(1 << i); + } + + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events 0-3 will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_component_driver ts3a227e_soc_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (ts3a227e == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..e2acf9c --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * 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. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif -- cgit v0.10.2 From f10c0a71e6efc7c8cbc3bfcfd0ecf822607f0b3d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:56 +0100 Subject: ASoC: tlv320aic32x4: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 6ea662d..015467e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -597,18 +597,6 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_suspend(struct snd_soc_codec *codec) -{ - aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int aic32x4_resume(struct snd_soc_codec *codec) -{ - aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); @@ -654,8 +642,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); - aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* * Workaround: for an unknown reason, the ADC needs to be powered up * and down for the first capture to work properly. It seems related to @@ -669,18 +655,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) return 0; } -static int aic32x4_remove(struct snd_soc_codec *codec) -{ - aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .probe = aic32x4_probe, - .remove = aic32x4_remove, - .suspend = aic32x4_suspend, - .resume = aic32x4_resume, .set_bias_level = aic32x4_set_bias_level, + .suspend_bias_off = true, .controls = aic32x4_snd_controls, .num_controls = ARRAY_SIZE(aic32x4_snd_controls), -- cgit v0.10.2 From 68f438378cde79e29f71c7e043b10d76001d8892 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:57 +0100 Subject: ASoC: tlv320aic3x: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f0a8281..b7ebce0 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1491,7 +1491,6 @@ static int aic3x_remove(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int i; - aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, -- cgit v0.10.2 From 90db15e17e46e2841843bf20f258fed963228bed Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:58 +0100 Subject: ASoC: tlv320dac33: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index e21ed93..0fe2ced 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1436,8 +1436,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); destroy_workqueue(dac33->dac33_wq); -- cgit v0.10.2 From 3ec8d2036464d961a6314281ae68d80a5c071d07 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:59 +0100 Subject: ASoC: twl4030: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b6b0cb3..27f3b21 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2177,8 +2177,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct twl4030_codec_data *pdata = twl4030->pdata; - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio)) gpio_free(pdata->hs_extmute_gpio); -- cgit v0.10.2