CHROMIUM: ASoC: Import entire upstream ASoC tree

sound/soc/pxa/magician.c

 /*
  * SoC audio for HTC Magician
  *
  * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
  *
  * based on spitz.c,
  * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  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.
  *
  */
 
 #include <linux/module.h>
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <linux/delay.h>
 #include <linux/gpio.h>
 #include <linux/i2c.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
-#include <sound/soc-dapm.h>
 #include <sound/uda1380.h>
 
 #include <mach/magician.h>
 #include <asm/mach-types.h>
 #include "../codecs/uda1380.h"
 #include "pxa2xx-i2s.h"
 #include "pxa-ssp.h"
 
 #define MAGICIAN_MIC       0
 #define MAGICIAN_MIC_EXT   1
 
 static int magician_hp_switch;
 static int magician_spk_switch = 1;
 static int magician_in_sel = MAGICIAN_MIC;
 
 static void magician_ext_control(struct snd_soc_codec *codec)
 {
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
+
         if (magician_spk_switch)
-                snd_soc_dapm_enable_pin(codec, "Speaker");
+                snd_soc_dapm_enable_pin(dapm, "Speaker");
         else
-                snd_soc_dapm_disable_pin(codec, "Speaker");
+                snd_soc_dapm_disable_pin(dapm, "Speaker");
         if (magician_hp_switch)
-                snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+                snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
         else
-                snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+                snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
 
         switch (magician_in_sel) {
         case MAGICIAN_MIC:
-                snd_soc_dapm_disable_pin(codec, "Headset Mic");
-                snd_soc_dapm_enable_pin(codec, "Call Mic");
+                snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+                snd_soc_dapm_enable_pin(dapm, "Call Mic");
                 break;
         case MAGICIAN_MIC_EXT:
-                snd_soc_dapm_disable_pin(codec, "Call Mic");
-                snd_soc_dapm_enable_pin(codec, "Headset Mic");
+                snd_soc_dapm_disable_pin(dapm, "Call Mic");
+                snd_soc_dapm_enable_pin(dapm, "Headset Mic");
                 break;
         }
 
-        snd_soc_dapm_sync(codec);
+        snd_soc_dapm_sync(dapm);
 }
 
 static int magician_startup(struct snd_pcm_substream *substream)
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_codec *codec = rtd->codec;
 
         mutex_lock(&codec->mutex);
 
         /* check the jack status at stream startup */
@@ skipping 314 matching lines @@
         SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
                         magician_get_input, magician_set_input),
 };
 
 /*
  * Logic for a uda1380 as connected on a HTC Magician
  */
 static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
         struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
         int err;
 
         /* NC codec pins */
-        snd_soc_dapm_nc_pin(codec, "VOUTLHP");
-        snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+        snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
+        snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
 
         /* FIXME: is anything connected here? */
-        snd_soc_dapm_nc_pin(codec, "VINL");
-        snd_soc_dapm_nc_pin(codec, "VINR");
+        snd_soc_dapm_nc_pin(dapm, "VINL");
+        snd_soc_dapm_nc_pin(dapm, "VINR");
 
         /* Add magician specific controls */
         err = snd_soc_add_controls(codec, uda1380_magician_controls,
                                 ARRAY_SIZE(uda1380_magician_controls));
         if (err < 0)
                 return err;
 
         /* Add magician specific widgets */
-        snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+        snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
                                   ARRAY_SIZE(uda1380_dapm_widgets));
 
         /* Set up magician specific audio path interconnects */
-        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_sync(codec);
+        snd_soc_dapm_sync(dapm);
         return 0;
 }
 
 /* magician digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link magician_dai[] = {
 {
         .name = "uda1380",
         .stream_name = "UDA1380 Playback",
         .cpu_dai_name = "pxa-ssp-dai.0",
         .codec_dai_name = "uda1380-hifi-playback",
@@ skipping 117 matching lines @@
         gpio_free(EGPIO_MAGICIAN_EP_POWER);
         gpio_free(EGPIO_MAGICIAN_SPK_POWER);
 }
 
 module_init(magician_init);
 module_exit(magician_exit);
 
 MODULE_AUTHOR("Philipp Zabel");
 MODULE_DESCRIPTION("ALSA SoC Magician");
 MODULE_LICENSE("GPL");