Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-mpc5200' into asoc-next

Documentation/devicetree/bindings/sound/fsl-asoc-card.txt

@@ -24,6 +24,9 @@ The compatible list for this generic sound card currently:
 
  "fsl,imx-audio-cs42888"
 
+ "fsl,imx-audio-cs427x"
+ (compatible with CS4271 and CS4272)
+
  "fsl,imx-audio-wm8962"
  (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt)
 
@@ -63,6 +66,12 @@ Optional properties:
   - audio-asrc		: The phandle of ASRC. It can be absent if there's no
 			  need to add ASRC support via DPCM.
 
+Optional unless SSI is selected as a CPU DAI:
+
+  - mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX)
+
+  - mux-ext-port	: The external port of the i.MX audio muxer
+
 Example:
 sound-cs42888 {
 	compatible = "fsl,imx-audio-cs42888";

sound/soc/davinci/Kconfig

@@ -5,7 +5,7 @@ config SND_DAVINCI_SOC
 
 config SND_EDMA_SOC
 	tristate "SoC Audio for Texas Instruments chips using eDMA"
-	depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI
+	depends on TI_EDMA
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y or M here if you want audio support for TI SoC which uses eDMA.
@@ -13,6 +13,7 @@ config SND_EDMA_SOC
 	  - daVinci devices
 	  - AM335x
 	  - AM437x/AM438x
+	  - DRA7xx family
 
 config SND_DAVINCI_SOC_I2S
 	tristate

sound/soc/davinci/davinci-mcasp.c

@@ -77,6 +77,7 @@ struct davinci_mcasp {
 	u32 fifo_base;
 	struct device *dev;
 	struct snd_pcm_substream *substreams[2];
+	unsigned int dai_fmt;
 
 	/* McASP specific data */
 	int	tdm_slots;
@@ -398,6 +399,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	bool fs_pol_rising;
 	bool inv_fs = false;
 
+	if (!fmt)
+		return 0;
+
 	pm_runtime_get_sync(mcasp->dev);
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_A:
@@ -529,6 +533,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
 	}
+
+	mcasp->dai_fmt = fmt;
 out:
 	pm_runtime_put(mcasp->dev);
 	return ret;
@@ -1026,6 +1032,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	int period_size = params_period_size(params);
 	int ret;
 
+	ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt);
+	if (ret)
+		return ret;
+
 	/*
 	 * If mcasp is BCLK master, and a BCLK divider was not provided by
 	 * the machine driver, we need to calculate the ratio.
@@ -1517,6 +1527,8 @@ static int mcasp_reparent_fck(struct platform_device *pdev)
 	if (!parent_name)
 		return 0;
 
+	dev_warn(&pdev->dev, "Update the bindings to use assigned-clocks!\n");
+
 	gfclk = clk_get(&pdev->dev, "fck");
 	if (IS_ERR(gfclk)) {
 		dev_err(&pdev->dev, "failed to get fck\n");

sound/soc/fsl/Kconfig

@@ -292,8 +292,8 @@ config SND_SOC_FSL_ASOC_CARD
 	select SND_SOC_FSL_SSI
 	help
 	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
-	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
-	 and SGTL5000.
+	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888,
+	 CS4271, CS4272 and SGTL5000.
 	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
 
 endif # SND_IMX_SOC

sound/soc/fsl/fsl-asoc-card.c

@@ -28,6 +28,8 @@
 #include "../codecs/wm8962.h"
 #include "../codecs/wm8960.h"
 
+#define CS427x_SYSCLK_MCLK 0
+
 #define RX 0
 #define TX 1
 
@@ -99,19 +101,26 @@ struct fsl_asoc_card_priv {
 /**
  * This dapm route map exsits for DPCM link only.
  * The other routes shall go through Device Tree.
+ *
+ * Note: keep all ASRC routes in the second half
+ *	 to drop them easily for non-ASRC cases.
  */
 static const struct snd_soc_dapm_route audio_map[] = {
-	{"CPU-Playback",  NULL, "ASRC-Playback"},
+	/* 1st half -- Normal DAPM routes */
 	{"Playback",  NULL, "CPU-Playback"},
-	{"ASRC-Capture",  NULL, "CPU-Capture"},
 	{"CPU-Capture",  NULL, "Capture"},
+	/* 2nd half -- ASRC DAPM routes */
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+	{"ASRC-Capture",  NULL, "CPU-Capture"},
 };
 
 static const struct snd_soc_dapm_route audio_map_ac97[] = {
-	{"AC97 Playback",  NULL, "ASRC-Playback"},
+	/* 1st half -- Normal DAPM routes */
 	{"Playback",  NULL, "AC97 Playback"},
-	{"ASRC-Capture",  NULL, "AC97 Capture"},
 	{"AC97 Capture",  NULL, "Capture"},
+	/* 2nd half -- ASRC DAPM routes */
+	{"AC97 Playback",  NULL, "ASRC-Playback"},
+	{"ASRC-Capture",  NULL, "AC97 Capture"},
 };
 
 /* Add all possible widgets into here without being redundant */
@@ -528,6 +537,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
 		priv->cpu_priv.slot_width = 32;
 		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
+		codec_dai_name = "cs4271-hifi";
+		priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
 	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
 		codec_dai_name = "sgtl5000";
 		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
@@ -593,6 +606,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
 
+	/* Drop the second half of DAPM routes -- ASRC */
+	if (!asrc_pdev)
+		priv->card.num_dapm_routes /= 2;
+
 	memcpy(priv->dai_link, fsl_asoc_card_dai,
 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
 
@@ -681,6 +698,7 @@ fail:
 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-ac97", },
 	{ .compatible = "fsl,imx-audio-cs42888", },
+	{ .compatible = "fsl,imx-audio-cs427x", },
 	{ .compatible = "fsl,imx-audio-sgtl5000", },
 	{ .compatible = "fsl,imx-audio-wm8962", },
 	{ .compatible = "fsl,imx-audio-wm8960", },

sound/soc/fsl/mpc5200_psc_ac97.c

@@ -13,6 +13,7 @@
 #include <linux/of_device.h>
 #include <linux/of_platform.h>
 #include <linux/delay.h>
+#include <linux/time.h>
 
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -127,7 +128,7 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
 
 	mutex_unlock(&psc_dma->mutex);
 
-	msleep(1);
+	usleep_range(1000, 2000);
 	psc_ac97_warm_reset(ac97);
 }