Merge remote-tracking branches 'asoc/topic/tas5086', 'asoc/topic/tegra',... (84b720f0) · Commits · e / devices / android_kernel_oneplus_sm7250

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt

0 → 100644

+42 −0

Original line number	Diff line number	Diff line
		NVIDIA Tegra audio complex, with SGTL5000 CODEC

		Required properties:
		- compatible : "nvidia,tegra-audio-sgtl5000"
		- clocks : Must contain an entry for each entry in clock-names.
		See ../clocks/clock-bindings.txt for details.
		- clock-names : Must include the following entries:
		- pll_a
		- pll_a_out0
		- mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
		- nvidia,model : The user-visible name of this sound complex.
		- nvidia,audio-routing : A list of the connections between audio components.
		Each entry is a pair of strings, the first being the connection's sink,
		the second being the connection's source. Valid names for sources and
		sinks are the SGTL5000's pins (as documented in its binding), and the jacks
		on the board:

		* Headphone Jack
		* Line In Jack
		* Mic Jack

		- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
		connected to the CODEC.
		- nvidia,audio-codec : The phandle of the SGTL5000 audio codec.

		Example:

		sound {
		compatible = "toradex,tegra-audio-sgtl5000-apalis_t30",
		"nvidia,tegra-audio-sgtl5000";
		nvidia,model = "Toradex Apalis T30";
		nvidia,audio-routing =
		"Headphone Jack", "HP_OUT",
		"LINE_IN", "Line In Jack",
		"MIC_IN", "Mic Jack";
		nvidia,i2s-controller = <&tegra_i2s2>;
		nvidia,audio-codec = <&sgtl5000>;
		clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
		<&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
		<&tegra_car TEGRA30_CLK_EXTERN1>;
		clock-names = "pll_a", "pll_a_out0", "mclk";
		};

Documentation/devicetree/bindings/sound/tlv320aic31xx.txt

+6 −3

Original line number	Diff line number	Diff line
		@@ -11,6 +11,7 @@ Required properties:
		"ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
		"ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
		"ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
		"ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP)

		- reg - <int> - I2C slave address
		- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
		@@ -37,9 +38,11 @@ CODEC output pins:
		* MICBIAS

		CODEC input pins:
		* MIC1LP
		* MIC1RP
		* MIC1LM
		* MIC1LP, devices with ADC
		* MIC1RP, devices with ADC
		* MIC1LM, devices with ADC
		* AIN1, devices without ADC
		* AIN2, devices without ADC

		The pins can be used in referring sound node's audio-routing property.

include/uapi/sound/asoc.h

+33 −2

Original line number	Diff line number	Diff line
		@@ -83,7 +83,7 @@
		#define SND_SOC_TPLG_NUM_TEXTS 16

		/* ABI version */
		#define SND_SOC_TPLG_ABI_VERSION 0x4
		#define SND_SOC_TPLG_ABI_VERSION 0x5

		/* Max size of TLV data */
		#define SND_SOC_TPLG_TLV_SIZE 32
		@@ -105,7 +105,8 @@
		#define SND_SOC_TPLG_TYPE_CODEC_LINK 9
		#define SND_SOC_TPLG_TYPE_BACKEND_LINK 10
		#define SND_SOC_TPLG_TYPE_PDATA 11
		#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA
		#define SND_SOC_TPLG_TYPE_BE_DAI 12
		#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_BE_DAI

		/* vendor block IDs - please add new vendor types to end */
		#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000
		@@ -124,6 +125,11 @@
		#define SND_SOC_TPLG_TUPLE_TYPE_WORD 4
		#define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5

		/* BE DAI flags */
		#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES (1 << 0)
		#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1)
		#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2)

		/*
		* Block Header.
		* This header precedes all object and object arrays below.
		@@ -251,6 +257,7 @@ struct snd_soc_tplg_stream_caps {
		__le32 period_size_max; /* max period size bytes */
		__le32 buffer_size_min; /* min buffer size bytes */
		__le32 buffer_size_max; /* max buffer size bytes */
		__le32 sig_bits; /* number of bits of content */
		} __attribute__((packed));

		/*
		@@ -285,6 +292,8 @@ struct snd_soc_tplg_manifest {
		__le32 graph_elems; /* number of graph elements */
		__le32 pcm_elems; /* number of PCM elements */
		__le32 dai_link_elems; /* number of DAI link elements */
		__le32 be_dai_elems; /* number of BE DAI elements */
		__le32 reserved[20]; /* reserved for new ABI element types */
		struct snd_soc_tplg_private priv;
		} __attribute__((packed));

		@@ -450,4 +459,26 @@ struct snd_soc_tplg_link_config {
		struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
		__le32 num_streams; /* number of streams */
		} __attribute__((packed));

		/*
		* Describes SW/FW specific features of BE DAI.
		*
		* File block representation for BE DAI :-
		* +-----------------------------------+-----+
		* \| struct snd_soc_tplg_hdr \| 1 \|
		* +-----------------------------------+-----+
		* \| struct snd_soc_tplg_be_dai \| N \|
		* +-----------------------------------+-----+
		*/
		struct snd_soc_tplg_be_dai {
		__le32 size; /* in bytes of this structure */
		char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */
		__le32 dai_id; /* unique ID - used to match */
		__le32 playback; /* supports playback mode */
		__le32 capture; /* supports capture mode */
		struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */
		__le32 flag_mask; /* bitmask of flags to configure */
		__le32 flags; /* SND_SOC_TPLG_DAI_FLGBIT_* */
		struct snd_soc_tplg_private priv;
		} __attribute__((packed));
		#endif

sound/soc/codecs/tas5086.c

+1 −1

Original line number	Diff line number	Diff line
		@@ -387,7 +387,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream,
		val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios),
		priv->mclk / priv->rate);
		if (val < 0) {
		dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n");
		dev_err(codec->dev, "Invalid MCLK / Fs ratio\n");
		return -EINVAL;
		}

sound/soc/codecs/tlv320aic31xx.c

+158 −58

Original line number	Diff line number	Diff line
		@@ -273,10 +273,20 @@ static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
		/*
		* controls to be exported to the user space
		*/
		static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
		static const struct snd_kcontrol_new common31xx_snd_controls[] = {
		SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
		AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),

		SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
		AIC31XX_HPRGAIN, 2, 1, 0),
		SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
		AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),

		SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
		AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
		};

		static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
		SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
		adc_fgain_tlv),

		@@ -286,14 +296,6 @@ static const struct snd_kcontrol_new aic31xx_snd_controls[] = {

		SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
		119, 0, mic_pga_tlv),

		SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
		AIC31XX_HPRGAIN, 2, 1, 0),
		SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
		AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),

		SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
		AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
		};

		static const struct snd_kcontrol_new aic311x_snd_controls[] = {
		@@ -397,17 +399,28 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
		return 0;
		}

		static const struct snd_kcontrol_new left_output_switches[] = {
		static const struct snd_kcontrol_new aic31xx_left_output_switches[] = {
		SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
		SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
		SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
		};

		static const struct snd_kcontrol_new right_output_switches[] = {
		static const struct snd_kcontrol_new aic31xx_right_output_switches[] = {
		SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
		SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
		};

		static const struct snd_kcontrol_new dac31xx_left_output_switches[] = {
		SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
		SOC_DAPM_SINGLE("From AIN1", AIC31XX_DACMIXERROUTE, 5, 1, 0),
		SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 4, 1, 0),
		};

		static const struct snd_kcontrol_new dac31xx_right_output_switches[] = {
		SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
		SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 1, 1, 0),
		};

		static const struct snd_kcontrol_new p_term_mic1lp =
		SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);

		@@ -457,7 +470,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
		return 0;
		}

		static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
		static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = {
		SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),

		SND_SOC_DAPM_MUX("DAC Left Input",
		@@ -473,14 +486,7 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
		AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
		SND_SOC_DAPM_POST_PMU \| SND_SOC_DAPM_POST_PMD),

		/* Output Mixers */
		SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
		left_output_switches,
		ARRAY_SIZE(left_output_switches)),
		SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
		right_output_switches,
		ARRAY_SIZE(right_output_switches)),

		/* HP */
		SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
		&aic31xx_dapm_hpl_switch),
		SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
		@@ -494,10 +500,34 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
		NULL, 0, aic31xx_dapm_power_event,
		SND_SOC_DAPM_POST_PMD \| SND_SOC_DAPM_POST_PMU),

		/* ADC */
		SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
		aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU \|
		SND_SOC_DAPM_POST_PMD),
		/* Mic Bias */
		SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
		SND_SOC_DAPM_POST_PMU \| SND_SOC_DAPM_PRE_PMD),

		/* Outputs */
		SND_SOC_DAPM_OUTPUT("HPL"),
		SND_SOC_DAPM_OUTPUT("HPR"),
		};

		static const struct snd_soc_dapm_widget dac31xx_dapm_widgets[] = {
		/* Inputs */
		SND_SOC_DAPM_INPUT("AIN1"),
		SND_SOC_DAPM_INPUT("AIN2"),

		/* Output Mixers */
		SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
		dac31xx_left_output_switches,
		ARRAY_SIZE(dac31xx_left_output_switches)),
		SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
		dac31xx_right_output_switches,
		ARRAY_SIZE(dac31xx_right_output_switches)),
		};

		static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
		/* Inputs */
		SND_SOC_DAPM_INPUT("MIC1LP"),
		SND_SOC_DAPM_INPUT("MIC1RP"),
		SND_SOC_DAPM_INPUT("MIC1LM"),

		/* Input Selection to MIC_PGA */
		SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
		@@ -507,24 +537,25 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
		SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
		&p_term_mic1lm),

		/* ADC */
		SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
		aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU \|
		SND_SOC_DAPM_POST_PMD),

		SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
		&m_term_mic1lm),

		/* Enabling & Disabling MIC Gain Ctl */
		SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
		7, 1, NULL, 0),

		/* Mic Bias */
		SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
		SND_SOC_DAPM_POST_PMU \| SND_SOC_DAPM_PRE_PMD),

		/* Outputs */
		SND_SOC_DAPM_OUTPUT("HPL"),
		SND_SOC_DAPM_OUTPUT("HPR"),

		/* Inputs */
		SND_SOC_DAPM_INPUT("MIC1LP"),
		SND_SOC_DAPM_INPUT("MIC1RP"),
		SND_SOC_DAPM_INPUT("MIC1LM"),
		/* Output Mixers */
		SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
		aic31xx_left_output_switches,
		ARRAY_SIZE(aic31xx_left_output_switches)),
		SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
		aic31xx_right_output_switches,
		ARRAY_SIZE(aic31xx_right_output_switches)),
		};

		static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
		@@ -554,7 +585,7 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
		};

		static const struct snd_soc_dapm_route
		aic31xx_audio_map[] = {
		common31xx_audio_map[] = {
		/* DAC Input Routing */
		{"DAC Left Input", "Left Data", "DAC IN"},
		{"DAC Left Input", "Right Data", "DAC IN"},
		@@ -565,6 +596,31 @@ aic31xx_audio_map[] = {
		{"DAC Left", NULL, "DAC Left Input"},
		{"DAC Right", NULL, "DAC Right Input"},

		/* HPL path */
		{"HP Left", "Switch", "Output Left"},
		{"HPL Driver", NULL, "HP Left"},
		{"HPL", NULL, "HPL Driver"},

		/* HPR path */
		{"HP Right", "Switch", "Output Right"},
		{"HPR Driver", NULL, "HP Right"},
		{"HPR", NULL, "HPR Driver"},
		};

		static const struct snd_soc_dapm_route
		dac31xx_audio_map[] = {
		/* Left Output */
		{"Output Left", "From Left DAC", "DAC Left"},
		{"Output Left", "From AIN1", "AIN1"},
		{"Output Left", "From AIN2", "AIN2"},

		/* Right Output */
		{"Output Right", "From Right DAC", "DAC Right"},
		{"Output Right", "From AIN2", "AIN2"},
		};

		static const struct snd_soc_dapm_route
		aic31xx_audio_map[] = {
		/* Mic input */
		{"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
		{"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
		@@ -595,16 +651,6 @@ aic31xx_audio_map[] = {
		/* Right Output */
		{"Output Right", "From Right DAC", "DAC Right"},
		{"Output Right", "From MIC1RP", "MIC1RP"},

		/* HPL path */
		{"HP Left", "Switch", "Output Left"},
		{"HPL Driver", NULL, "HP Left"},
		{"HPL", NULL, "HPL Driver"},

		/* HPR path */
		{"HP Right", "Switch", "Output Right"},
		{"HPR Driver", NULL, "HP Right"},
		{"HPR", NULL, "HPR Driver"},
		};

		static const struct snd_soc_dapm_route
		@@ -633,6 +679,13 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec)
		int ret = 0;
		struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);

		if (!(aic31xx->pdata.codec_type & DAC31XX_BIT))
		ret = snd_soc_add_codec_controls(
		codec, aic31xx_snd_controls,
		ARRAY_SIZE(aic31xx_snd_controls));
		if (ret)
		return ret;

		if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT)
		ret = snd_soc_add_codec_controls(
		codec, aic311x_snd_controls,
		@@ -651,6 +704,30 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec)
		struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
		int ret = 0;

		if (aic31xx->pdata.codec_type & DAC31XX_BIT) {
		ret = snd_soc_dapm_new_controls(
		dapm, dac31xx_dapm_widgets,
		ARRAY_SIZE(dac31xx_dapm_widgets));
		if (ret)
		return ret;

		ret = snd_soc_dapm_add_routes(dapm, dac31xx_audio_map,
		ARRAY_SIZE(dac31xx_audio_map));
		if (ret)
		return ret;
		} else {
		ret = snd_soc_dapm_new_controls(
		dapm, aic31xx_dapm_widgets,
		ARRAY_SIZE(aic31xx_dapm_widgets));
		if (ret)
		return ret;

		ret = snd_soc_dapm_add_routes(dapm, aic31xx_audio_map,
		ARRAY_SIZE(aic31xx_audio_map));
		if (ret)
		return ret;
		}

		if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
		ret = snd_soc_dapm_new_controls(
		dapm, aic311x_dapm_widgets,
		@@ -1115,12 +1192,12 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
		.suspend_bias_off = true,

		.component_driver = {
		.controls = aic31xx_snd_controls,
		.num_controls = ARRAY_SIZE(aic31xx_snd_controls),
		.dapm_widgets = aic31xx_dapm_widgets,
		.num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets),
		.dapm_routes = aic31xx_audio_map,
		.num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map),
		.controls = common31xx_snd_controls,
		.num_controls = ARRAY_SIZE(common31xx_snd_controls),
		.dapm_widgets = common31xx_dapm_widgets,
		.num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets),
		.dapm_routes = common31xx_audio_map,
		.num_dapm_routes = ARRAY_SIZE(common31xx_audio_map),
		},
		};

		@@ -1131,19 +1208,34 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = {
		.digital_mute = aic31xx_dac_mute,
		};

		static struct snd_soc_dai_driver dac31xx_dai_driver[] = {
		{
		.name = "tlv32dac31xx-hifi",
		.playback = {
		.stream_name = "Playback",
		.channels_min = 2,
		.channels_max = 2,
		.rates = AIC31XX_RATES,
		.formats = AIC31XX_FORMATS,
		},
		.ops = &aic31xx_dai_ops,
		.symmetric_rates = 1,
		}
		};

		static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
		{
		.name = "tlv320aic31xx-hifi",
		.playback = {
		.stream_name = "Playback",
		.channels_min = 1,
		.channels_min = 2,
		.channels_max = 2,
		.rates = AIC31XX_RATES,
		.formats = AIC31XX_FORMATS,
		},
		.capture = {
		.stream_name = "Capture",
		.channels_min = 1,
		.channels_min = 2,
		.channels_max = 2,
		.rates = AIC31XX_RATES,
		.formats = AIC31XX_FORMATS,
		@@ -1261,7 +1353,14 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
		if (ret)
		return ret;

		return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
		if (aic31xx->pdata.codec_type & DAC31XX_BIT)
		return snd_soc_register_codec(&i2c->dev,
		&soc_codec_driver_aic31xx,
		dac31xx_dai_driver,
		ARRAY_SIZE(dac31xx_dai_driver));
		else
		return snd_soc_register_codec(&i2c->dev,
		&soc_codec_driver_aic31xx,
		aic31xx_dai_driver,
		ARRAY_SIZE(aic31xx_dai_driver));
		}
		@@ -1279,6 +1378,7 @@ static const struct i2c_device_id aic31xx_i2c_id[] = {
		{ "tlv320aic3110", AIC3110 },
		{ "tlv320aic3120", AIC3120 },
		{ "tlv320aic3111", AIC3111 },
		{ "tlv320dac3100", DAC3100 },
		{ }
		};
		MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);