Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
  ALSA: ASoC codec: remove unused #include <version.h>
  ALSA: ASoC: update email address for Liam Girdwood
  ALSA: hda: corrected invalid mixer values
  ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
  ALSA: ASoC: Add destination and source port for DMA on OMAP1
  ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
  ALSA: ASoC: Fix build of GTA01 audio driver
  ALSA: ASoC: Add widgets before setting endpoints on GTA01
  ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
  ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
  ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
  ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
  ALSA: ASoC: Make TLV320AIC26 user-visible
  ALSA: ASoC - clean up Kconfig for TLV320AIC2
  ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
  ALSA: ASoC: Implement WM8510 bias level control
  ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
  ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
  ALSA: ASoC: Add WM8510 SPI support
  ALSA: ASoC: Add WM8753 SPI support
  ...

1 files changed, 232 insertions, 0 deletions

diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 00000000000..0fe73379689
--- /dev/null
+++ b/sound/soc/omap/osk5912.c
@@ -0,0 +1,232 @@
+/*
+ * osk5912.c  --  SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS  <arunks@mistralsolutions.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+	return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_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->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return err;
+	}
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return err;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	err =
+	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return err;
+	}
+
+	return err;
+}
+
+static struct snd_soc_ops osk_ops = {
+	.startup = osk_startup,
+	.hw_params = osk_hw_params,
+	.shutdown = osk_shutdown,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic Jack"},
+};
+
+static int osk_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add osk5912 specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up osk5912 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = osk_tlv320aic23_init,
+	.ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_osk = {
+	.name = "OSK5912",
+	.dai_link = &osk_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+	.machine = &snd_soc_machine_osk,
+	.platform = &omap_soc_platform,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+	int err;
+	u32 curRate;
+	struct device *dev;
+
+	if (!(machine_is_omap_osk()))
+		return -ENODEV;
+
+	osk_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!osk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
+	osk_snd_devdata.dev = &osk_snd_device->dev;
+	*(unsigned int *)osk_dai.cpu_dai->private_data = 0;	/* McBSP1 */
+	err = platform_device_add(osk_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &osk_snd_device->dev;
+
+	tlv320aic23_mclk = clk_get(dev, "mclk");
+	if (IS_ERR(tlv320aic23_mclk)) {
+		printk(KERN_ERR "Could not get mclk clock\n");
+		return -ENODEV;
+	}
+
+	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
+		/* MCLK is already in use */
+		printk(KERN_WARNING
+		       "MCLK in use at %d Hz. We change it to %d Hz\n",
+		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+	}
+
+	/*
+	 * Configure 12 MHz output on MCLK.
+	 */
+	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+	if (curRate != CODEC_CLOCK) {
+		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+			err = -ECANCELED;
+			goto err1;
+		}
+	}
+
+	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
+	       clk_get_usecount(tlv320aic23_mclk));
+
+	return 0;
+err1:
+	clk_put(tlv320aic23_mclk);
+	platform_device_del(osk_snd_device);
+	platform_device_put(osk_snd_device);
+
+	return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+	platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");