Merge branch 'fix/asoc' into for-linus

* fix/asoc:
  ASoC: Fix warning in wm9705
  ASoC: OMAP: Update contact addresses
  ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR

diff --combined sound/soc/omap/omap-pcm.c
index 1bdbb04,b078ed5..07cf7f4
--- 1/sound/soc/omap/omap-pcm.c
--- 2/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@@ -3,7 -3,8 +3,8 @@@
   *
   * Copyright (C) 2008 Nokia Corporation
   *
-  * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+  * Contact: Jarkko Nikula <jhnikula@gmail.com>
+  *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
   *
   * This program is free software; you can redistribute it and/or
   * modify it under the terms of the GNU General Public License
@@@ -327,7 -328,7 +328,7 @@@ int omap_pcm_new(struct snd_card *card
        if (!card->dev->dma_mask)
                card->dev->dma_mask = &omap_pcm_dmamask;
        if (!card->dev->coherent_dma_mask)
 -              card->dev->coherent_dma_mask = DMA_32BIT_MASK;
 +              card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
  
        if (dai->playback.channels_min) {
                ret = omap_pcm_preallocate_dma_buffer(pcm,
@@@ -367,6 -368,6 +368,6 @@@ static void __exit omap_soc_platform_ex
  }
  module_exit(omap_soc_platform_exit);
  
- MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+ MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
  MODULE_DESCRIPTION("OMAP PCM DMA module");
  MODULE_LICENSE("GPL");

diff --combined sound/soc/pxa/pxa-ssp.c
index de22544,74ff69e..286be31
--- 1/sound/soc/pxa/pxa-ssp.c
--- 2/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@@ -30,7 -30,7 +30,7 @@@
  #include <sound/pxa2xx-lib.h>
  
  #include <mach/hardware.h>
 -#include <mach/pxa-regs.h>
 +#include <mach/dma.h>
  #include <mach/regs-ssp.h>
  #include <mach/audio.h>
  #include <mach/ssp.h>
@@@ -280,12 -280,33 +280,33 @@@ static int pxa_ssp_resume(struct snd_so
   * ssp_set_clkdiv - set SSP clock divider
   * @div: serial clock rate divider
   */
- static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+ static void ssp_set_scr(struct ssp_device *ssp, u32 div)
  {
-       struct ssp_device *ssp = dev->ssp;
-       u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+       u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+ 
+       if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+               sscr0 &= ~0x0000ff00;
+               sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+       } else {
+               sscr0 &= ~0x000fff00;
+               sscr0 |= (div - 1) << 8;     /* 1..4096 */
+       }
+       ssp_write_reg(ssp, SSCR0, sscr0);
+ }
+ 
+ /**
+  * ssp_get_clkdiv - get SSP clock divider
+  */
+ static u32 ssp_get_scr(struct ssp_device *ssp)
+ {
+       u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+       u32 div;
  
-       ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+       if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+               div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+       else
+               div = ((sscr0 >> 8) & 0xfff) + 1;
+       return div;
  }
  
  /*
@@@ -326,7 -347,7 +347,7 @@@ static int pxa_ssp_set_dai_sysclk(struc
                break;
        case PXA_SSP_CLK_AUDIO:
                priv->sysclk = 0;
-               ssp_set_scr(&priv->dev, 1);
+               ssp_set_scr(ssp, 1);
                sscr0 |= SSCR0_ACS;
                break;
        default:
@@@ -387,7 -408,7 +408,7 @@@ static int pxa_ssp_set_dai_clkdiv(struc
                ssp_write_reg(ssp, SSACD, val);
                break;
        case PXA_SSP_DIV_SCR:
-               ssp_set_scr(&priv->dev, div);
+               ssp_set_scr(ssp, div);
                break;
        default:
                return -ENODEV;
@@@ -674,8 -695,7 +695,7 @@@ static int pxa_ssp_hw_params(struct snd
        case SND_SOC_DAIFMT_I2S:
               sspsp = ssp_read_reg(ssp, SSPSP);
  
-               if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
-                    (width == 16)) {
+               if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
                        /* This is a special case where the bitclk is 64fs
                        * and we're not dealing with 2*32 bits of audio
                        * samples.