CHROMIUM: ASoC: Import entire upstream ASoC tree

sound/soc/jz4740/qi_lb60.c

 /*
  * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
  *
  * 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.
  *
  *  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.,
  *  675 Mass Ave, Cambridge, MA 02139, USA.
  *
  */
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
-#include <sound/soc-dapm.h>
 #include <linux/gpio.h>
 
 #define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29)
 #define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4)
 
 static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget,
                              struct snd_kcontrol *ctrl, int event)
 {
         int on = 0;
         if (event & SND_SOC_DAPM_POST_PMU)
@@ skipping 19 matching lines @@
 };
 
 #define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \
                         SND_SOC_DAIFMT_NB_NF | \
                         SND_SOC_DAIFMT_CBM_CFM)
 
 static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
         struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
         int ret;
 
-        snd_soc_dapm_nc_pin(codec, "LIN");
-        snd_soc_dapm_nc_pin(codec, "RIN");
+        snd_soc_dapm_nc_pin(dapm, "LIN");
+        snd_soc_dapm_nc_pin(dapm, "RIN");
 
         ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
         if (ret < 0) {
                 dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret);
                 return ret;
         }
 
-        snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets));
-        snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes));
-        snd_soc_dapm_sync(codec);
+        snd_soc_dapm_new_controls(dapm, qi_lb60_widgets,
+                                  ARRAY_SIZE(qi_lb60_widgets));
+        snd_soc_dapm_add_routes(dapm, qi_lb60_routes,
+                                ARRAY_SIZE(qi_lb60_routes));
+        snd_soc_dapm_sync(dapm);
 
         return 0;
 }
 
 static struct snd_soc_dai_link qi_lb60_dai = {
         .name = "jz4740",
         .stream_name = "jz4740",
         .cpu_dai_name = "jz4740-i2s",
         .platform_name = "jz4740-pcm-audio",
         .codec_dai_name = "jz4740-hifi",
@@ skipping 62 matching lines @@
 {
         gpio_free(QI_LB60_AMP_GPIO);
         gpio_free(QI_LB60_SND_GPIO);
         platform_device_unregister(qi_lb60_snd_device);
 }
 module_exit(qi_lb60_exit);
 
 MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
 MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support");
 MODULE_LICENSE("GPL v2");