Merge remote-tracking branches 'asoc/topic/rt5651', 'asoc/topic/samsu…

…ng', 'asoc/topic/sgtl5000', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf', 'asoc/topic/sta350' and 'asoc/topic/tlv320dac33' into asoc-next

Documentation/devicetree/bindings/sound/simple-card.txt

@@ -1,6 +1,6 @@
 Simple-Card:
 
-Simple-Card specifies audio DAI connection of SoC <-> codec.
+Simple-Card specifies audio DAI connections of SoC <-> codec.
 
 Required properties:
 
@@ -10,56 +10,73 @@ Optional properties:
 
 - simple-audio-card,name		: User specified audio sound card name, one string
 					  property.
-- simple-audio-card,format		: CPU/CODEC common audio format.
-					  "i2s", "right_j", "left_j" , "dsp_a"
-					  "dsp_b", "ac97", "pdm", "msb", "lsb"
 - simple-audio-card,widgets		: Please refer to widgets.txt.
 - simple-audio-card,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.
-- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
-- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
+Optional subnodes:
+
+- simple-audio-card,dai-link		: Container for dai-link level
+					  properties and the CPU and CODEC
+					  sub-nodes. This container may be
+					  omitted when the card has only one
+					  DAI link. See the examples and the
+					  section bellow.
+
+Dai-link subnode properties and subnodes:
+
+If dai-link subnode is omitted and the subnode properties are directly
+under "sound"-node the subnode property and subnode names have to be
+prefixed with "simple-audio-card,"-prefix.
 
-Required subnodes:
+Required dai-link subnodes:
 
-- simple-audio-card,dai-link		: container for the CPU and CODEC sub-nodes
-					  This container may be omitted when the
-					  card has only one DAI link.
-					  See the examples.
+- cpu					: CPU   sub-node
+- codec					: CODEC sub-node
 
-- simple-audio-card,cpu			: CPU   sub-node
-- simple-audio-card,codec		: CODEC sub-node
+Optional dai-link subnode properties:
+
+- format				: CPU/CODEC common audio format.
+					  "i2s", "right_j", "left_j" , "dsp_a"
+					  "dsp_b", "ac97", "pdm", "msb", "lsb"
+- frame-master				: Indicates dai-link frame master.
+					  phandle to a cpu or codec subnode.
+- bitclock-master			: Indicates dai-link bit clock master.
+					  phandle to a cpu or codec subnode.
+- bitclock-inversion			: bool property. Add this if the
+					  dai-link uses bit clock inversion.
+- frame-inversion			: bool property. Add this if the
+					  dai-link uses frame clock inversion.
+
+For backward compatibility the frame-master and bitclock-master
+properties can be used as booleans in codec subnode to indicate if the
+codec is the dai-link frame or bit clock master. In this case there
+should be no dai-link node, the same properties should not be present
+at sound-node level, and the bitclock-inversion and frame-inversion
+properties should also be placed in the codec node if needed.
 
 Required CPU/CODEC subnodes properties:
 
 - sound-dai				: phandle and port of CPU/CODEC
 
 Optional CPU/CODEC subnodes properties:
 
-- format				: CPU/CODEC specific audio format if needed.
-					  see simple-audio-card,format
-- frame-master				: bool property. add this if subnode is frame master
-- bitclock-master			: bool property. add this if subnode is bitclock master
-- bitclock-inversion			: bool property. add this if subnode has clock inversion
-- frame-inversion			: bool property. add this if subnode has frame inversion
+- dai-tdm-slot-num			: Please refer to tdm-slot.txt.
+- dai-tdm-slot-width			: Please refer to tdm-slot.txt.
 - clocks / system-clock-frequency	: specify subnode's clock if needed.
 					  it can be specified via "clocks" if system has
 					  clock node (= common clock), or "system-clock-frequency"
 					  (if system doens't support common clock)
 
-Note:
- * For 'format', 'frame-master', 'bitclock-master', 'bitclock-inversion' and
-   'frame-inversion', the simple card will use the settings of CODEC for both
-   CPU and CODEC sides as we need to keep the settings identical for both ends
-   of the link.
-
 Example 1 - single DAI link:
 
 sound {
 	compatible = "simple-audio-card";
 	simple-audio-card,name = "VF610-Tower-Sound-Card";
 	simple-audio-card,format = "left_j";
+	simple-audio-card,bitclock-master = <&dailink0_master>;
+	simple-audio-card,frame-master = <&dailink0_master>;
 	simple-audio-card,widgets =
 		"Microphone", "Microphone Jack",
 		"Headphone", "Headphone Jack",
@@ -69,17 +86,12 @@ sound {
 		"Headphone Jack", "HP_OUT",
 		"External Speaker", "LINE_OUT";
 
-	dai-tdm-slot-num = <2>;
-	dai-tdm-slot-width = <8>;
-
 	simple-audio-card,cpu {
 		sound-dai = <&sh_fsi2 0>;
 	};
 
-	simple-audio-card,codec {
+	dailink0_master: simple-audio-card,codec {
 		sound-dai = <&ak4648>;
-		bitclock-master;
-		frame-master;
 		clocks = <&osc>;
 	};
 };
@@ -105,31 +117,31 @@ Example 2 - many DAI links:
 sound {
 	compatible = "simple-audio-card";
 	simple-audio-card,name = "Cubox Audio";
-	simple-audio-card,format = "i2s";
 
 	simple-audio-card,dai-link@0 {		/* I2S - HDMI */
-		simple-audio-card,cpu {
+		format = "i2s";
+		cpu {
 			sound-dai = <&audio1 0>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&tda998x 0>;
 		};
 	};
 
 	simple-audio-card,dai-link@1 {		/* S/PDIF - HDMI */
-		simple-audio-card,cpu {
+		cpu {
 			sound-dai = <&audio1 1>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&tda998x 1>;
 		};
 	};
 
 	simple-audio-card,dai-link@2 {		/* S/PDIF - S/PDIF */
-		simple-audio-card,cpu {
+		cpu {
 			sound-dai = <&audio1 1>;
 		};
-		simple-audio-card,codec {
+		codec {
 			sound-dai = <&spdif_codec>;
 		};
 	};

Documentation/devicetree/bindings/sound/snow.txt

@@ -0,0 +1,17 @@
+Audio Binding for Snow boards
+
+Required properties:
+- compatible : Can be one of the following,
+			"google,snow-audio-max98090" or
+			"google,snow-audio-max98095"
+- samsung,i2s-controller: The phandle of the Samsung I2S controller
+- samsung,audio-codec: The phandle of the audio codec
+
+Example:
+
+sound {
+		compatible = "google,snow-audio-max98095";
+
+		samsung,i2s-controller = <&i2s0>;
+		samsung,audio-codec = <&max98095>;
+};

Documentation/devicetree/bindings/sound/st,sta350.txt

@@ -25,6 +25,7 @@ Optional properties:
 	2: 2 Channel (Full-Bridge) Power, 1 Channel FFX
 	3: 1 Channel Mono-Parallel
 	If parameter is missing, mode 0 will be enabled.
+	This property has to be specified as '/bits/ 8' value.
 
   -  st,ch1-output-mapping: Channel 1 output mapping
   -  st,ch2-output-mapping: Channel 2 output mapping
@@ -33,6 +34,7 @@ Optional properties:
 	1: Channel 2
 	2: Channel 3
 	If parameter is missing, channel 1 is choosen.
+	This properties have to be specified as '/bits/ 8' values.
 
   -  st,thermal-warning-recover:
 	If present, thermal warning recovery is enabled.
@@ -82,9 +84,31 @@ Optional properties:
 	If not present, preset DC coefficient is used.
 
   -  st,invalid-input-detect-mute:
-	If not present, automatic invalid input detect mute is enabled.
+	If present, automatic invalid input detect mute is enabled.
 
+  -  st,activate-mute-output:
+	If present, a mute output will be activated in ase the volume will
+	reach a value lower than -76 dBFS.
 
+  -  st,bridge-immediate-off:
+	If present, the bridge will be switched off immediately after the
+	power-down-gpio goes low. Otherwise, the bridge will wait for 13
+	million clock cycles to pass before shutting down.
+
+  -  st,noise-shape-dc-cut:
+	If present, the noise-shaping technique on the DC cutoff filter are
+	enabled.
+
+  -  st,powerdown-master-volume:
+	If present, the power-down pin and I2C power-down functions will
+	act on the master volume. Otherwise, the functions will act on the
+	mute commands.
+
+  -  st,powerdown-delay-divider:
+	If present, the bridge power-down time will be divided by the provided
+	value. If not specified, a divider of 1 will be used. Allowed values
+	are 1, 2, 4, 8, 16, 32, 64 and 128.
+	This property has to be specified as '/bits/ 8' value.
 
 Example:
 
@@ -93,12 +117,12 @@ codec: sta350@38 {
 	reg = <0x1c>;
 	reset-gpios = <&gpio1 19 0>;
 	power-down-gpios = <&gpio1 16 0>;
-	st,output-conf = <0x3>;			// set output to 2-channel
+	st,output-conf = /bits/ 8  <0x3>;	// set output to 2-channel
 						// (full-bridge) power,
 						// 2-channel data-out
-	st,ch1-output-mapping = <0>;		// set channel 1 output ch 1
-	st,ch2-output-mapping = <0>;		// set channel 2 output ch 1
-	st,ch3-output-mapping = <0>;		// set channel 3 output ch 1
+	st,ch1-output-mapping = /bits/ 8 <0>;	// set channel 1 output ch 1
+	st,ch2-output-mapping = /bits/ 8 <0>;	// set channel 2 output ch 1
+	st,ch3-output-mapping = /bits/ 8 <0>;	// set channel 3 output ch 1
 	st,max-power-correction;		// enables power bridge
 						// correction for THD reduction
 						// near maximum power output

include/sound/soc.h

@@ -1326,7 +1326,9 @@ int snd_soc_of_parse_tdm_slot(struct device_node *np,
 int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 				   const char *propname);
 unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
-				     const char *prefix);
+				     const char *prefix,
+				     struct device_node **bitclkmaster,
+				     struct device_node **framemaster);
 int snd_soc_of_get_dai_name(struct device_node *of_node,
 			    const char **dai_name);
 

include/sound/sta350.h

@@ -37,6 +37,7 @@ struct sta350_platform_data {
 	u8 ch3_output_mapping;
 	u8 ffx_power_output_mode;
 	u8 drop_compensation_ns;
+	u8 powerdown_delay_divider;
 	unsigned int thermal_warning_recovery:1;
 	unsigned int thermal_warning_adjustment:1;
 	unsigned int fault_detect_recovery:1;
@@ -47,6 +48,10 @@ struct sta350_platform_data {
 	unsigned int odd_pwm_speed_mode:1;
 	unsigned int distortion_compensation:1;
 	unsigned int invalid_input_detect_mute:1;
+	unsigned int activate_mute_output:1;
+	unsigned int bridge_immediate_off:1;
+	unsigned int noise_shape_dc_cut:1;
+	unsigned int powerdown_master_vol:1;
 };
 
 #endif /* __LINUX_SND__STA350_H */

sound/soc/codecs/rt5651.c

@@ -1368,8 +1368,7 @@ static int get_clk_info(int sclk, int rate)
 static int rt5651_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_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec);
 	unsigned int val_len = 0, val_clk, mask_clk;
 	int pre_div, bclk_ms, frame_size;

sound/soc/codecs/sgtl5000.c

@@ -1322,24 +1322,21 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
 			return ret;
 	}
 
-	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+	ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
 				 sgtl5000->supplies);
 	if (ret)
 		goto err_ldo_remove;
 
 	ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
 					sgtl5000->supplies);
 	if (ret)
-		goto err_regulator_free;
+		goto err_ldo_remove;
 
 	/* wait for all power rails bring up */
 	udelay(10);
 
 	return 0;
 
-err_regulator_free:
-	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
 err_ldo_remove:
 	if (!external_vddd)
 		ldo_regulator_remove(codec);
@@ -1409,8 +1406,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 err:
 	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
-	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
 	ldo_regulator_remove(codec);
 
 	return ret;
@@ -1424,8 +1419,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
 
 	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
-	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
-				sgtl5000->supplies);
 	ldo_regulator_remove(codec);
 
 	return 0;