Merge remote-tracking branches 'asoc/topic/rx51', 'asoc/topic/samsung',...

Merge remote-tracking branches 'asoc/topic/rx51', 'asoc/topic/samsung', 'asoc/topic/sh', 'asoc/topic/simple' and 'asoc/topic/sta32x' into asoc-next

Documentation/devicetree/bindings/sound/samsung-i2s.txt

@@ -33,6 +33,25 @@ Required SoC Specific Properties:
   "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root
   clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2
   doesn't have any such mux.
+- #clock-cells: should be 1, this property must be present if the I2S device
+  is a clock provider in terms of the common clock bindings, described in
+  ../clock/clock-bindings.txt.
+- clock-output-names: from the common clock bindings, names of the CDCLK
+  I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1",
+  "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices recpectively.
+
+There are following clocks available at the I2S device nodes:
+ CLK_I2S_CDCLK    - the CDCLK (CODECLKO) gate clock,
+ CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the
+		    IISPSR register),
+ CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in
+		    IISMOD register).
+
+Refer to the SoC datasheet for availability of the above clocks.
+The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available
+in the IIS Multi Audio Interface (I2S0).
+Note: Old DTs may not have the #clock-cells, clock-output-names properties
+and then not use the I2S node as a clock supplier.
 
 Optional SoC Specific Properties:
 
@@ -41,6 +60,7 @@ Optional SoC Specific Properties:
 - pinctrl-0: Should specify pin control groups used for this controller.
 - pinctrl-names: Should contain only one value - "default".
 
+
 Example:
 
 i2s0: i2s@03830000 {
@@ -54,6 +74,8 @@ i2s0: i2s@03830000 {
 		<&clock_audss EXYNOS_I2S_BUS>,
 		<&clock_audss EXYNOS_SCLK_I2S>;
 	clock-names = "iis", "i2s_opclk0", "i2s_opclk1";
+	#clock-cells;
+	clock-output-names = "i2s_cdclk0";
 	samsung,idma-addr = <0x03000000>;
 	pinctrl-names = "default";
 	pinctrl-0 = <&i2s0_bus>;

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

@@ -75,6 +75,11 @@ Optional CPU/CODEC subnodes properties:
 					  it can be specified via "clocks" if system has
 					  clock node (= common clock), or "system-clock-frequency"
 					  (if system doens't support common clock)
+					  If a clock is specified, it is
+					  enabled with clk_prepare_enable()
+					  in dai startup() and disabled with
+					  clk_disable_unprepare() in dai
+					  shutdown().
 
 Example 1 - single DAI link:
 

Documentation/devicetree/bindings/sound/st,sta32x.txt

+STA32X audio CODEC
+
+The driver for this device only supports I2C.
+
+Required properties:
+
+  - compatible: "st,sta32x"
+  - reg: the I2C address of the device for I2C
+  - reset-gpios: a GPIO spec for the reset pin. If specified, it will be
+		 deasserted before communication to the codec starts.
+
+  - power-down-gpios: a GPIO spec for the power down pin. If specified,
+		      it will be deasserted before communication to the codec
+		      starts.
+
+  - Vdda-supply: regulator spec, providing 3.3V
+  - Vdd3-supply: regulator spec, providing 3.3V
+  - Vcc-supply: regulator spec, providing 5V - 26V
+
+Optional properties:
+
+  -  st,output-conf: number, Selects the output configuration:
+	0: 2-channel (full-bridge) power, 2-channel data-out
+	1: 2 (half-bridge). 1 (full-bridge) on-board power
+	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
+  -  st,ch3-output-mapping: Channel 3 output mapping
+	0: Channel 1
+	1: Channel 2
+	2: Channel 3
+	If parameter is missing, channel 1 is chosen.
+	This properties have to be specified as '/bits/ 8' values.
+
+  -  st,thermal-warning-recover:
+	If present, thermal warning recovery is enabled.
+
+  -  st,thermal-warning-adjustment:
+	If present, thermal warning adjustment is enabled.
+
+  -  st,fault-detect-recovery:
+	If present, then fault recovery will be enabled.
+
+  -  st,drop-compensation-ns: number
+	Only required for "st,ffx-power-output-mode" ==
+	"variable-drop-compensation".
+	Specifies the drop compensation in nanoseconds.
+	The value must be in the range of 0..300, and only
+	multiples of 20 are allowed. Default is 140ns.
+
+  -  st,max-power-use-mpcc:
+	If present, then MPCC bits are used for MPC coefficients,
+	otherwise standard MPC coefficients are used.
+
+  -  st,max-power-corr:
+	If present, power bridge correction for THD reduction near maximum
+	power output is enabled.
+
+  -  st,am-reduction-mode:
+	If present, FFX mode runs in AM reduction mode, otherwise normal
+	FFX mode is used.
+
+  -  st,odd-pwm-speed-mode:
+	If present, PWM speed mode run on odd speed mode (341.3 kHz) on all
+	channels. If not present, normal PWM spped mode (384 kHz) will be used.
+
+  -  st,invalid-input-detect-mute:
+	If present, automatic invalid input detect mute is enabled.
+
+Example:
+
+codec: sta32x@38 {
+	compatible = "st,sta32x";
+	reg = <0x1c>;
+	reset-gpios = <&gpio1 19 0>;
+	power-down-gpios = <&gpio1 16 0>;
+	st,output-conf = /bits/ 8  <0x3>;	// set output to 2-channel
+						// (full-bridge) power,
+						// 2-channel data-out
+	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
+	st,invalid-input-detect-mute;		// mute if no valid digital
+						// audio signal is provided.
+};

include/dt-bindings/sound/samsung-i2s.h

+#ifndef _DT_BINDINGS_SAMSUNG_I2S_H
+#define _DT_BINDINGS_SAMSUNG_I2S_H
+
+#define CLK_I2S_CDCLK		0
+#define CLK_I2S_RCLK_SRC	1
+#define CLK_I2S_RCLK_PSR	2
+
+#endif /* _DT_BINDINGS_SAMSUNG_I2S_H */

include/sound/simple_card.h

@@ -20,6 +20,7 @@ struct asoc_simple_dai {
 	unsigned int sysclk;
 	int slots;
 	int slot_width;
+	struct clk *clk;
 };
 
 struct asoc_simple_card_info {

include/sound/sta32x.h

@@ -24,12 +24,20 @@
 #define STA32X_THERMAL_RECOVERY_ENABLE		2
 
 struct sta32x_platform_data {
-	int output_conf;
-	int ch1_output_mapping;
-	int ch2_output_mapping;
-	int ch3_output_mapping;
-	int thermal_conf;
+	u8 output_conf;
+	u8 ch1_output_mapping;
+	u8 ch2_output_mapping;
+	u8 ch3_output_mapping;
 	int needs_esd_watchdog;
+	u8 drop_compensation_ns;
+	unsigned int thermal_warning_recovery:1;
+	unsigned int thermal_warning_adjustment:1;
+	unsigned int fault_detect_recovery:1;
+	unsigned int max_power_use_mpcc:1;
+	unsigned int max_power_correction:1;
+	unsigned int am_reduction_mode:1;
+	unsigned int odd_pwm_speed_mode:1;
+	unsigned int invalid_input_detect_mute:1;
 };
 
 #endif /* __LINUX_SND__STA32X_H */

sound/soc/codecs/Kconfig

@@ -580,7 +580,9 @@ config SND_SOC_SSM4567
 	depends on I2C
 
 config SND_SOC_STA32X
-	tristate
+	tristate "STA326, STA328 and STA329 speaker amplifier"
+	depends on I2C
+	select REGMAP_I2C
 
 config SND_SOC_STA350
 	tristate "STA350 speaker amplifier"

sound/soc/generic/simple-card.c

@@ -39,6 +39,37 @@ struct simple_card_data {
 #define simple_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i)
 #define simple_priv_to_props(priv, i) ((priv)->dai_props + i)
 
+static int asoc_simple_card_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct simple_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
+	struct simple_dai_props *dai_props =
+		&priv->dai_props[rtd - rtd->card->rtd];
+	int ret;
+
+	ret = clk_prepare_enable(dai_props->cpu_dai.clk);
+	if (ret)
+		return ret;
+	
+	ret = clk_prepare_enable(dai_props->codec_dai.clk);
+	if (ret)
+		clk_disable_unprepare(dai_props->cpu_dai.clk);
+
+	return ret;
+}
+
+static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct simple_card_data *priv =	snd_soc_card_get_drvdata(rtd->card);
+	struct simple_dai_props *dai_props =
+		&priv->dai_props[rtd - rtd->card->rtd];
+
+	clk_disable_unprepare(dai_props->cpu_dai.clk);
+
+	clk_disable_unprepare(dai_props->codec_dai.clk);
+}
+
 static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream,
 				      struct snd_pcm_hw_params *params)
 {
@@ -58,6 +89,8 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream,
 }
 
 static struct snd_soc_ops asoc_simple_card_ops = {
+	.startup = asoc_simple_card_startup,
+	.shutdown = asoc_simple_card_shutdown,
 	.hw_params = asoc_simple_card_hw_params,
 };
 
@@ -219,6 +252,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
 		}
 
 		dai->sysclk = clk_get_rate(clk);
+		dai->clk = clk;
 	} else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
 		dai->sysclk = val;
 	} else {

sound/soc/omap/rx51.c

@@ -250,14 +250,14 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"FM Transmitter", NULL, "LLOUT"},
 	{"FM Transmitter", NULL, "RLOUT"},
 
-	{"DMic Rate 64", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "DMic"},
+	{"DMic Rate 64", NULL, "DMic"},
+	{"DMic", NULL, "Mic Bias"},
 
 	{"b LINE2R", NULL, "MONO_LOUT"},
 	{"Earphone", NULL, "b HPLOUT"},
 
-	{"LINE1L", NULL, "b Mic Bias"},
-	{"b Mic Bias", NULL, "HS Mic"}
+	{"LINE1L", NULL, "HS Mic"},
+	{"HS Mic", NULL, "b Mic Bias"},
 };
 
 static const char * const spk_function[] = {"Off", "On"};

sound/soc/samsung/Kconfig

@@ -54,7 +54,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
 config SND_SOC_SAMSUNG_SMDK_WM8580
 	tristate "SoC I2S Audio support for WM8580 on SMDK"
 	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
-	depends on REGMAP_I2C
+	depends on I2C
 	select SND_SOC_WM8580
 	select SND_SAMSUNG_I2S
 	help
@@ -167,7 +167,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF
 config SND_SOC_SMDK_WM8580_PCM
 	tristate "SoC PCM Audio support for WM8580 on SMDK"
 	depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110)
-	depends on REGMAP_I2C
+	depends on I2C
 	select SND_SOC_WM8580
 	select SND_SAMSUNG_PCM
 	help

sound/soc/samsung/arndale_rt5631.c

@@ -135,7 +135,6 @@ MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match);
 static struct platform_driver arndale_audio_driver = {
 	.driver = {
 		.name   = "arndale-audio",
-		.owner  = THIS_MODULE,
 		.pm = &snd_soc_pm_ops,
 		.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
 	},

sound/soc/samsung/jive_wm8750.c

@@ -83,22 +83,6 @@ static struct snd_soc_ops jive_ops = {
 	.hw_params	= jive_hw_params,
 };
 
-static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	/* These endpoints are not being used. */
-	snd_soc_dapm_nc_pin(dapm, "LINPUT2");
-	snd_soc_dapm_nc_pin(dapm, "RINPUT2");
-	snd_soc_dapm_nc_pin(dapm, "LINPUT3");
-	snd_soc_dapm_nc_pin(dapm, "RINPUT3");
-	snd_soc_dapm_nc_pin(dapm, "OUT3");
-	snd_soc_dapm_nc_pin(dapm, "MONO");
-
-	return 0;
-}
-
 static struct snd_soc_dai_link jive_dai = {
 	.name		= "wm8750",
 	.stream_name	= "WM8750",
@@ -106,7 +90,6 @@ static struct snd_soc_dai_link jive_dai = {
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name	= "s3c2412-i2s",
 	.codec_name	= "wm8750.0-001a",
-	.init		= jive_wm8750_init,
 	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
 			  SND_SOC_DAIFMT_CBS_CFS,
 	.ops		= &jive_ops,
@@ -123,6 +106,7 @@ static struct snd_soc_card snd_soc_machine_jive = {
 	.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
 	.dapm_routes	= audio_map,
 	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.fully_routed	= true,
 };
 
 static struct platform_device *jive_snd_device;

sound/soc/samsung/odroidx2_max98090.c

@@ -21,6 +21,8 @@ struct odroidx2_drv_data {
 /* The I2S CDCLK output clock frequency for the MAX98090 codec */
 #define MAX98090_MCLK 19200000
 
+static struct snd_soc_dai_link odroidx2_dai[];
+
 static int odroidx2_late_probe(struct snd_soc_card *card)
 {
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
@@ -29,7 +31,9 @@ static int odroidx2_late_probe(struct snd_soc_card *card)
 
 	ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK,
 						SND_SOC_CLOCK_IN);
-	if (ret < 0)
+
+	if (ret < 0 || of_find_property(odroidx2_dai[0].codec_of_node,
+					"clocks", NULL))
 		return ret;
 
 	/* Set the cpu DAI configuration in order to use CDCLK */

sound/soc/samsung/smdk_wm8580.c

@@ -136,13 +136,10 @@ static const struct snd_soc_dapm_route smdk_wm8580_audio_map[] = {
 
 static int smdk_wm8580_init_paiftx(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
 	/* Enabling the microphone requires the fitting of a 0R
 	 * resistor to connect the line from the microphone jack.
 	 */
-	snd_soc_dapm_disable_pin(dapm, "MicIn");
+	snd_soc_dapm_disable_pin(&rtd->card->dapm, "MicIn");
 
 	return 0;
 }

sound/soc/sh/dma-sh7760.c

@@ -305,11 +305,6 @@ static struct snd_pcm_ops camelot_pcm_ops = {
 	.pointer	= camelot_pos,
 };
 
-static void camelot_pcm_free(struct snd_pcm *pcm)
-{
-	snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
 static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_pcm *pcm = rtd->pcm;
@@ -328,7 +323,6 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_platform_driver sh7760_soc_platform = {
 	.ops		= &camelot_pcm_ops,
 	.pcm_new	= camelot_pcm_new,
-	.pcm_free	= camelot_pcm_free,
 };
 
 static int sh7760_soc_platform_probe(struct platform_device *pdev)

sound/soc/sh/fsi.c

@@ -1762,11 +1762,6 @@ static struct snd_pcm_ops fsi_pcm_ops = {
 #define PREALLOC_BUFFER		(32 * 1024)
 #define PREALLOC_BUFFER_MAX	(32 * 1024)
 
-static void fsi_pcm_free(struct snd_pcm *pcm)
-{
-	snd_pcm_lib_preallocate_free_for_all(pcm);
-}
-
 static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
 	return snd_pcm_lib_preallocate_pages_for_all(
@@ -1818,7 +1813,6 @@ static struct snd_soc_dai_driver fsi_soc_dai[] = {
 static struct snd_soc_platform_driver fsi_soc_platform = {
 	.ops		= &fsi_pcm_ops,
 	.pcm_new	= fsi_pcm_new,
-	.pcm_free	= fsi_pcm_free,
 };
 
 static const struct snd_soc_component_driver fsi_soc_component = {

sound/soc/sh/siu_pcm.c

@@ -589,7 +589,6 @@ static void siu_pcm_free(struct snd_pcm *pcm)
 	tasklet_kill(&port_info->playback.tasklet);
 
 	siu_free_port(port_info);
-	snd_pcm_lib_preallocate_free_for_all(pcm);
 
 	dev_dbg(pcm->card->dev, "%s\n", __func__);
 }