CHROMIUM: ASoC: Import entire upstream ASoC tree

sound/soc/omap/osk5912.c

 /*
  * 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 <plat/mcbsp.h>
 
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 #include "../codecs/tlv320aic23.h"
 
@@ skipping 69 matching lines @@
 
         {"LLINEIN", NULL, "Line In"},
         {"RLINEIN", NULL, "Line In"},
 
         {"MICIN", NULL, "Mic Jack"},
 };
 
 static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
 
         /* Add osk5912 specific widgets */
-        snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+        snd_soc_dapm_new_controls(dapm, 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_add_routes(dapm, 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_enable_pin(dapm, "Headphone Jack");
+        snd_soc_dapm_enable_pin(dapm, "Line In");
+        snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
-        snd_soc_dapm_sync(codec);
+        snd_soc_dapm_sync(dapm);
 
         return 0;
 }
 
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link osk_dai = {
         .name = "TLV320AIC23",
         .stream_name = "AIC23",
         .cpu_dai_name = "omap-mcbsp-dai.0",
         .codec_dai_name = "tlv320aic23-hifi",
@@ skipping 72 matching lines @@
         clk_put(tlv320aic23_mclk);
         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");