Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "A relative calm release at this time with a flat diffstat.  The only
  significant change in the ALSA core side is the support for more than
  32 card instances, configurable via kconfig.  Other than that, in both
  ASoC and other parts, mostly some improvements and fixes on the driver
  side.

   - hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
   - hda: Haswell HDMI audio fixes, runtime PM improvements
   - hda: Intel BayTrail support, ALC5505 DSP support
   - es1968: MediaForte M56VAP support
   - usb-audio: Improved support for Yamaha/Roland devices
   - usb-audio: M2Tech hiFace, Audio Advantage Micro II support
   - hdspm: wordclock fixes
   - ASoC: Pending fixes for WM8962
   - ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
   - ASoC: Generalisation of the Bluetooth and HDMI stub drivers
   - ASoC: SSM2518 and RT5640 codec drivers.
   - ASoC: Tegra CPUs with RT5640 machine driver
   - ASoC: AC'97 refactoring bug fixes
   - ASoC: ADAU1701 driver fixes
   - Clean up of *_set_drvdata() in a wide range of drivers"

* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits)
  ALSA: vmaster: Fix the regression of missing vmaster hook call
  ALSA: hda - Add Dell SSID to support Headset Mic recording
  ASoC: adau1701: remove control_data assignment
  ASoC: adau1701: more direct regmap usage
  ASoC: ac97: fixup multi-platform AC'97 module build failure
  ASoC: pxa2xx: fixup multi-platform AC'97 build failures
  ASoC: tegra20-ac97: Remove unused variable
  ASoC: tegra20-ac97: Remove duplicate error message
  ALSA: usb-audio: Add Audio Advantage Micro II
  ASoC: tas5086: fix Mid-Z implementation
  ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
  ALSA: Replace the magic number 44 with const
  ALSA: hda - Fix the max length of control name in generic parser
  ALSA: hda - Guess what, it's two more Dell headset mic quirks
  ALSA: hda - Yet another Dell headset mic quirk
  ALSA: hda - Add support for ALC5505 DSP power-save mode
  ASoC: mfld: Remove unused variable
  ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
  ALSA: usb-audio: claim autodetected PCM interfaces all at once
  ALSA: usb-audio: remove superfluous Roland quirks
  ...

1 files changed, 323 insertions, 0 deletions

diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
new file mode 100644
index 000000000000..52a36a90f4f4
--- /dev/null
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -0,0 +1,323 @@
+/*
+ * Copyright 2013 Freescale Semiconductor, Inc.
+ *
+ * Based on imx-sgtl5000.c
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/of_i2c.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <linux/pinctrl/consumer.h>
+
+#include "../codecs/wm8962.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE	32
+
+struct imx_wm8962_data {
+	struct snd_soc_dai_link dai;
+	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;
+};
+static struct imx_priv card_priv;
+
+static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("AMIC", NULL),
+	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);
+
+	return 0;
+}
+
+static struct snd_soc_ops imx_hifi_ops = {
+	.hw_params = imx_hifi_hw_params,
+};
+
+static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct imx_priv *priv = &card_priv;
+	struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+	struct device *dev = &priv->pdev->dev;
+	unsigned int pll_out;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	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;
+			else
+				pll_out = sample_rate * 256;
+
+			ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+					WM8962_FLL_MCLK, data->clk_frequency,
+					pll_out);
+			if (ret < 0) {
+				dev_err(dev, "failed to start FLL: %d\n", ret);
+				return ret;
+			}
+
+			ret = snd_soc_dai_set_sysclk(codec_dai,
+					WM8962_SYSCLK_FLL, pll_out,
+					SND_SOC_CLOCK_IN);
+			if (ret < 0) {
+				dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+				return ret;
+			}
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (dapm->bias_level == SND_SOC_BIAS_PREPARE) {
+			ret = snd_soc_dai_set_sysclk(codec_dai,
+					WM8962_SYSCLK_MCLK, data->clk_frequency,
+					SND_SOC_CLOCK_IN);
+			if (ret < 0) {
+				dev_err(dev,
+					"failed to switch away from FLL: %d\n",
+					ret);
+				return ret;
+			}
+
+			ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+					0, 0, 0);
+			if (ret < 0) {
+				dev_err(dev, "failed to stop FLL: %d\n", ret);
+				return ret;
+			}
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	dapm->bias_level = level;
+
+	return 0;
+}
+
+static int imx_wm8962_late_probe(struct snd_soc_card *card)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct imx_priv *priv = &card_priv;
+	struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+	struct device *dev = &priv->pdev->dev;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+			data->clk_frequency, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(dev, "failed to set sysclk in %s\n", __func__);
+
+	return ret;
+}
+
+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;
+	int int_port, ext_port;
+	int ret;
+
+	priv->pdev = pdev;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+		return ret;
+	}
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+		return ret;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the audmux API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+	ret = imx_audmux_v2_configure_port(int_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(&pdev->dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+	imx_audmux_v2_configure_port(ext_port,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(&pdev->dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+	if (!ssi_np || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ssi_pdev = of_find_device_by_node(ssi_np);
+	if (!ssi_pdev) {
+		dev_err(&pdev->dev, "failed to find SSI platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+	codec_dev = of_find_i2c_device_by_node(codec_np);
+	if (!codec_dev || !codec_dev->driver) {
+		dev_err(&pdev->dev, "failed to find codec platform device\n");
+		return -EINVAL;
+	}
+
+	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto fail;
+	}
+
+	data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+	if (IS_ERR(data->codec_clk)) {
+		ret = PTR_ERR(data->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->dai.name = "HiFi";
+	data->dai.stream_name = "HiFi";
+	data->dai.codec_dai_name = "wm8962";
+	data->dai.codec_of_node = codec_np;
+	data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev);
+	data->dai.platform_of_node = ssi_np;
+	data->dai.ops = &imx_hifi_ops;
+	data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			    SND_SOC_DAIFMT_CBM_CFM;
+
+	data->card.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret)
+		goto clk_fail;
+	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+	if (ret)
+		goto clk_fail;
+	data->card.num_links = 1;
+	data->card.dai_link = &data->dai;
+	data->card.dapm_widgets = imx_wm8962_dapm_widgets;
+	data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
+
+	data->card.late_probe = imx_wm8962_late_probe;
+	data->card.set_bias_level = imx_wm8962_set_bias_level;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		goto clk_fail;
+	}
+
+	platform_set_drvdata(pdev, data);
+	of_node_put(ssi_np);
+	of_node_put(codec_np);
+
+	return 0;
+
+clk_fail:
+	if (!IS_ERR(data->codec_clk))
+		clk_disable_unprepare(data->codec_clk);
+fail:
+	if (ssi_np)
+		of_node_put(ssi_np);
+	if (codec_np)
+		of_node_put(codec_np);
+
+	return ret;
+}
+
+static int imx_wm8962_remove(struct platform_device *pdev)
+{
+	struct imx_wm8962_data *data = platform_get_drvdata(pdev);
+
+	if (!IS_ERR(data->codec_clk))
+		clk_disable_unprepare(data->codec_clk);
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_wm8962_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-wm8962", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids);
+
+static struct platform_driver imx_wm8962_driver = {
+	.driver = {
+		.name = "imx-wm8962",
+		.owner = THIS_MODULE,
+		.of_match_table = imx_wm8962_dt_ids,
+	},
+	.probe = imx_wm8962_probe,
+	.remove = imx_wm8962_remove,
+};
+module_platform_driver(imx_wm8962_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-wm8962");