Merge remote-tracking branches 'asoc/topic/fsl-ssi', 'asoc/topic/hi62…

…20' and 'asoc/topic/imx' into asoc-next

Documentation/devicetree/bindings/sound/fsl,ssi.txt

@@ -20,24 +20,8 @@ Required properties:
                     have.
 - interrupt-parent: The phandle for the interrupt controller that
                     services interrupts for this device.
-- fsl,playback-dma: Phandle to a node for the DMA channel to use for
-                    playback of audio.  This is typically dictated by SOC
-                    design.  See the notes below.
-- fsl,capture-dma:  Phandle to a node for the DMA channel to use for
-                    capture (recording) of audio.  This is typically dictated
-                    by SOC design.  See the notes below.
 - fsl,fifo-depth:   The number of elements in the transmit and receive FIFOs.
                     This number is the maximum allowed value for SFCSR[TFWM0].
-- fsl,ssi-asynchronous:
-                    If specified, the SSI is to be programmed in asynchronous
-                    mode.  In this mode, pins SRCK, STCK, SRFS, and STFS must
-                    all be connected to valid signals.  In synchronous mode,
-                    SRCK and SRFS are ignored.  Asynchronous mode allows
-                    playback and capture to use different sample sizes and
-                    sample rates.  Some drivers may require that SRCK and STCK
-                    be connected together, and SRFS and STFS be connected
-                    together.  This would still allow different sample sizes,
-                    but not different sample rates.
  - clocks:          "ipg" - Required clock for the SSI unit
                     "baud" - Required clock for SSI master mode. Otherwise this
 		      clock is not used
@@ -61,6 +45,24 @@ Optional properties:
 - fsl,mode:         The operating mode for the AC97 interface only.
                     "ac97-slave" - AC97 mode, SSI is clock slave
                     "ac97-master" - AC97 mode, SSI is clock master
+- fsl,ssi-asynchronous:
+                    If specified, the SSI is to be programmed in asynchronous
+                    mode.  In this mode, pins SRCK, STCK, SRFS, and STFS must
+                    all be connected to valid signals.  In synchronous mode,
+                    SRCK and SRFS are ignored.  Asynchronous mode allows
+                    playback and capture to use different sample sizes and
+                    sample rates.  Some drivers may require that SRCK and STCK
+                    be connected together, and SRFS and STFS be connected
+                    together.  This would still allow different sample sizes,
+                    but not different sample rates.
+- fsl,playback-dma: Phandle to a node for the DMA channel to use for
+                    playback of audio.  This is typically dictated by SOC
+                    design.  See the notes below.
+                    Only used on Power Architecture.
+- fsl,capture-dma:  Phandle to a node for the DMA channel to use for
+                    capture (recording) of audio.  This is typically dictated
+                    by SOC design.  See the notes below.
+                    Only used on Power Architecture.
 
 Child 'codec' node required properties:
 - compatible:       Compatible list, contains the name of the codec

Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt

@@ -0,0 +1,42 @@
+* Hisilicon 6210 i2s controller
+
+Required properties:
+
+- compatible: should be one of the following:
+   - "hisilicon,hi6210-i2s"
+- reg: physical base address of the i2s controller unit and length of
+   memory mapped region.
+- interrupts: should contain the i2s interrupt.
+- clocks: a list of phandle + clock-specifier pairs, one for each entry
+  in clock-names.
+- clock-names: should contain following:
+   - "dacodec"
+   - "i2s-base"
+- dmas: DMA specifiers for tx dma. See the DMA client binding,
+  Documentation/devicetree/bindings/dma/dma.txt
+- dma-names: should be "tx" and "rx"
+- hisilicon,sysctrl-syscon: phandle to sysctrl syscon
+- #sound-dai-cells: Should be set to 1 (for multi-dai)
+   - The dai cell indexes reference the following interfaces:
+       0: S2 interface
+       (Currently that is the only one available, but more may be
+        supported in the future)
+
+Example for the hi6210 i2s controller:
+
+i2s0: i2s@f7118000{
+	compatible = "hisilicon,hi6210-i2s";
+	reg = <0x0 0xf7118000 0x0 0x8000>; /* i2s unit */
+	interrupts = <GIC_SPI 123 IRQ_TYPE_LEVEL_HIGH>; /* 155 "DigACodec_intr"-32 */
+	clocks = <&sys_ctrl HI6220_DACODEC_PCLK>,
+		 <&sys_ctrl HI6220_BBPPLL0_DIV>;
+	clock-names = "dacodec", "i2s-base";
+	dmas = <&dma0 15 &dma0 14>;
+	dma-names = "rx", "tx";
+	hisilicon,sysctrl-syscon = <&sys_ctrl>;
+	#sound-dai-cells = <1>;
+};
+
+Then when referencing the i2s controller:
+	sound-dai = <&i2s0 0>; /* index 0 => S2 interface */
+

sound/soc/Kconfig

@@ -47,6 +47,7 @@ source "sound/soc/cirrus/Kconfig"
 source "sound/soc/davinci/Kconfig"
 source "sound/soc/dwc/Kconfig"
 source "sound/soc/fsl/Kconfig"
+source "sound/soc/hisilicon/Kconfig"
 source "sound/soc/jz4740/Kconfig"
 source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"

sound/soc/Makefile

@@ -27,6 +27,7 @@ obj-$(CONFIG_SND_SOC)	+= cirrus/
 obj-$(CONFIG_SND_SOC)	+= davinci/
 obj-$(CONFIG_SND_SOC)	+= dwc/
 obj-$(CONFIG_SND_SOC)	+= fsl/
+obj-$(CONFIG_SND_SOC)	+= hisilicon/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
 obj-$(CONFIG_SND_SOC)	+= img/
 obj-$(CONFIG_SND_SOC)	+= intel/

sound/soc/fsl/fsl_ssi.c

@@ -35,6 +35,7 @@
 #include <linux/module.h>
 #include <linux/interrupt.h>
 #include <linux/clk.h>
+#include <linux/ctype.h>
 #include <linux/device.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
@@ -54,16 +55,6 @@
 #include "fsl_ssi.h"
 #include "imx-pcm.h"
 
-/**
- * FSLSSI_I2S_RATES: sample rates supported by the I2S
- *
- * This driver currently only supports the SSI running in I2S slave mode,
- * which means the codec determines the sample rate.  Therefore, we tell
- * ALSA that we support all rates and let the codec driver decide what rates
- * are really supported.
- */
-#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS
-
 /**
  * FSLSSI_I2S_FORMATS: audio formats supported by the SSI
  *
@@ -1212,14 +1203,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 32,
-		.rates = FSLSSI_I2S_RATES,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 32,
-		.rates = FSLSSI_I2S_RATES,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
 	.ops = &fsl_ssi_dai_ops,
@@ -1325,14 +1316,10 @@ static struct snd_ac97_bus_ops fsl_ssi_ac97_ops = {
  */
 static void make_lowercase(char *s)
 {
-	char *p = s;
-	char c;
-
-	while ((c = *p)) {
-		if ((c >= 'A') && (c <= 'Z'))
-			*p = c + ('a' - 'A');
-		p++;
-	}
+	if (!s)
+		return;
+	for (; *s; s++)
+		*s = tolower(*s);
 }
 
 static int fsl_ssi_imx_probe(struct platform_device *pdev,

sound/soc/fsl/imx-pcm-dma.c

@@ -33,48 +33,20 @@ static bool filter(struct dma_chan *chan, void *param)
 	return true;
 }
 
-static const struct snd_pcm_hardware imx_pcm_hardware = {
-	.info = SNDRV_PCM_INFO_INTERLEAVED |
-		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP |
-		SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_PAUSE |
-		SNDRV_PCM_INFO_RESUME,
-	.buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE,
-	.period_bytes_min = 128,
-	.period_bytes_max = 65535, /* Limited by SDMA engine */
-	.periods_min = 2,
-	.periods_max = 255,
-	.fifo_size = 0,
-};
-
 static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = {
-	.pcm_hardware = &imx_pcm_hardware,
 	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
 	.compat_filter_fn = filter,
-	.prealloc_buffer_size = IMX_DEFAULT_DMABUF_SIZE,
 };
 
 int imx_pcm_dma_init(struct platform_device *pdev, size_t size)
 {
 	struct snd_dmaengine_pcm_config *config;
-	struct snd_pcm_hardware *pcm_hardware;
 
 	config = devm_kzalloc(&pdev->dev,
 			sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL);
 	if (!config)
 		return -ENOMEM;
 	*config = imx_dmaengine_pcm_config;
-	if (size)
-		config->prealloc_buffer_size = size;
-
-	pcm_hardware = devm_kzalloc(&pdev->dev,
-			sizeof(struct snd_pcm_hardware), GFP_KERNEL);
-	*pcm_hardware = imx_pcm_hardware;
-	if (size)
-		pcm_hardware->buffer_bytes_max = size;
-
-	config->pcm_hardware = pcm_hardware;
 
 	return devm_snd_dmaengine_pcm_register(&pdev->dev,
 		config,

sound/soc/fsl/imx-wm8962.c

@@ -33,14 +33,14 @@ struct imx_wm8962_data {
 	struct snd_soc_card card;
 	char codec_dai_name[DAI_NAME_SIZE];
 	char platform_name[DAI_NAME_SIZE];
-	struct clk *codec_clk;
 	unsigned int clk_frequency;
 };
 
 struct imx_priv {
 	struct platform_device *pdev;
+	int sample_rate;
+	snd_pcm_format_t sample_format;
 };
-static struct imx_priv card_priv;
 
 static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
@@ -49,14 +49,14 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
 	SND_SOC_DAPM_MIC("DMIC", NULL),
 };
 
-static int sample_rate = 44100;
-static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE;
-
 static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params)
 {
-	sample_rate = params_rate(params);
-	sample_format = params_format(params);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+
+	priv->sample_rate = params_rate(params);
+	priv->sample_format = params_format(params);
 
 	return 0;
 }
@@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 {
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai *codec_dai;
-	struct imx_priv *priv = &card_priv;
+	struct imx_priv *priv = snd_soc_card_get_drvdata(card);
 	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 	struct device *dev = &priv->pdev->dev;
 	unsigned int pll_out;
@@ -85,10 +85,10 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
 		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
-			if (sample_format == SNDRV_PCM_FORMAT_S24_LE)
-				pll_out = sample_rate * 384;
+			if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+				pll_out = priv->sample_rate * 384;
 			else
-				pll_out = sample_rate * 256;
+				pll_out = priv->sample_rate * 256;
 
 			ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
 					WM8962_FLL_MCLK, data->clk_frequency,
@@ -140,7 +140,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
 	struct snd_soc_dai *codec_dai;
-	struct imx_priv *priv = &card_priv;
+	struct imx_priv *priv = snd_soc_card_get_drvdata(card);
 	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 	struct device *dev = &priv->pdev->dev;
 	int ret;
@@ -160,13 +160,20 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *ssi_np, *codec_np;
 	struct platform_device *ssi_pdev;
-	struct imx_priv *priv = &card_priv;
 	struct i2c_client *codec_dev;
 	struct imx_wm8962_data *data;
+	struct imx_priv *priv;
+	struct clk *codec_clk;
 	int int_port, ext_port;
 	int ret;
 
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
 	priv->pdev = pdev;
+	priv->sample_rate = 44100;
+	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
 
 	ret = of_property_read_u32(np, "mux-int-port", &int_port);
 	if (ret) {
@@ -231,19 +238,15 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 		goto fail;
 	}
 
-	data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
-	if (IS_ERR(data->codec_clk)) {
-		ret = PTR_ERR(data->codec_clk);
+	codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (IS_ERR(codec_clk)) {
+		ret = PTR_ERR(codec_clk);
 		dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
 		goto fail;
 	}
 
-	data->clk_frequency = clk_get_rate(data->codec_clk);
-	ret = clk_prepare_enable(data->codec_clk);
-	if (ret) {
-		dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret);
-		goto fail;
-	}
+	data->clk_frequency = clk_get_rate(codec_clk);
+	clk_put(codec_clk);
 
 	data->dai.name = "HiFi";
 	data->dai.stream_name = "HiFi";
@@ -258,10 +261,10 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	data->card.dev = &pdev->dev;
 	ret = snd_soc_of_parse_card_name(&data->card, "model");
 	if (ret)
-		goto clk_fail;
+		goto fail;
 	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
 	if (ret)
-		goto clk_fail;
+		goto fail;
 	data->card.num_links = 1;
 	data->card.owner = THIS_MODULE;
 	data->card.dai_link = &data->dai;
@@ -277,34 +280,16 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
-		goto clk_fail;
+		goto fail;
 	}
 
-	of_node_put(ssi_np);
-	of_node_put(codec_np);
-
-	return 0;
-
-clk_fail:
-	clk_disable_unprepare(data->codec_clk);
 fail:
 	of_node_put(ssi_np);
 	of_node_put(codec_np);
 
 	return ret;
 }
 
-static int imx_wm8962_remove(struct platform_device *pdev)
-{
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
-	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
-
-	if (!IS_ERR(data->codec_clk))
-		clk_disable_unprepare(data->codec_clk);
-
-	return 0;
-}
-
 static const struct of_device_id imx_wm8962_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-wm8962", },
 	{ /* sentinel */ }
@@ -318,7 +303,6 @@ static struct platform_driver imx_wm8962_driver = {
 		.of_match_table = imx_wm8962_dt_ids,
 	},
 	.probe = imx_wm8962_probe,
-	.remove = imx_wm8962_remove,
 };
 module_platform_driver(imx_wm8962_driver);
 

sound/soc/hisilicon/Kconfig

@@ -0,0 +1,5 @@
+config SND_I2S_HI6210_I2S
+	tristate "Hisilicon I2S controller"
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+	help
+	  Hisilicon I2S

sound/soc/hisilicon/Makefile

@@ -0,0 +1 @@
+obj-$(CONFIG_SND_I2S_HI6210_I2S) += hi6210-i2s.o