Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...

1 files changed, 128 insertions, 0 deletions

diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c
new file mode 100644
index 000000000000..e01cb02abd3a
--- /dev/null
+++ b/sound/soc/cirrus/edb93xx.c
@@ -0,0 +1,128 @@
+/*
+ * SoC audio for EDB93xx
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include "ep93xx-pcm.h"
+
+static int edb93xx_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int err;
+	unsigned int mclk_rate;
+	unsigned int rate = params_rate(params);
+
+	/*
+	 * According to CS4271 datasheet we use MCLK/LRCK=256 for
+	 * rates below 50kHz and 128 for higher sample rates
+	 */
+	if (rate < 50000)
+		mclk_rate = rate * 64 * 4;
+	else
+		mclk_rate = rate * 64 * 2;
+
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
+				     SND_SOC_CLOCK_IN);
+	if (err)
+		return err;
+
+	return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
+				      SND_SOC_CLOCK_OUT);
+}
+
+static struct snd_soc_ops edb93xx_ops = {
+	.hw_params	= edb93xx_hw_params,
+};
+
+static struct snd_soc_dai_link edb93xx_dai = {
+	.name		= "CS4271",
+	.stream_name	= "CS4271 HiFi",
+	.platform_name	= "ep93xx-pcm-audio",
+	.cpu_dai_name	= "ep93xx-i2s",
+	.codec_name	= "spi0.0",
+	.codec_dai_name	= "cs4271-hifi",
+	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
+			  SND_SOC_DAIFMT_CBS_CFS,
+	.ops		= &edb93xx_ops,
+};
+
+static struct snd_soc_card snd_soc_edb93xx = {
+	.name		= "EDB93XX",
+	.owner		= THIS_MODULE,
+	.dai_link	= &edb93xx_dai,
+	.num_links	= 1,
+};
+
+static int __devinit edb93xx_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_edb93xx;
+	int ret;
+
+	ret = ep93xx_i2s_acquire();
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		ep93xx_i2s_release();
+	}
+
+	return ret;
+}
+
+static int __devexit edb93xx_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	ep93xx_i2s_release();
+
+	return 0;
+}
+
+static struct platform_driver edb93xx_driver = {
+	.driver		= {
+		.name	= "edb93xx-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= edb93xx_probe,
+	.remove		= __devexit_p(edb93xx_remove),
+};
+
+module_platform_driver(edb93xx_driver);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("ALSA SoC EDB93xx");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:edb93xx-audio");