Merge commit 'v2.6.30' into for-2.6.31

22 files changed, 233 insertions, 100 deletions

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
         .info             = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-                            SNDRV_PCM_INFO_INTERLEAVED,
+                            SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
         .formats          = AU1XPSC_PCM_FMTS,
         .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
         .period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..f2653803ede8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)    += snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)    += snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8990)    += snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991)    += snd-soc-wm8991.o
 obj-$(CONFIG_SND_SOC_WM9705)    += snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)    += snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)    += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
                 4, 3, 0, output_tvl),
 
         SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-                TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+                TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
         /* Common capture gain controls */
         SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..0275321ff8ab 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
          * required for LRC in master mode. The DACs or ADCs need a
          * valid audio path i.e. pin -> ADC or DAC -> pin before
          * the LRC will be enabled in master mode. */
-        if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+        if (!master || cmd != SNDRV_PCM_TRIGGER_START)
                 return 0;
 
         if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..9f6be3d31ac0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
                          struct snd_ctl_elem_value *ucontrol)
 {
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-        int reg = kcontrol->private_value & 0xff;
-        int reg2 = (kcontrol->private_value >> 24) & 0xff;
+        unsigned int reg = mc->reg;
+        unsigned int reg2 = mc->rreg;
         int ret;
         u16 val;
 
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
         return 0;
 }
 
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+                                    xinvert, tlv_array)                 \
 {       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
         .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
                 SNDRV_CTL_ELEM_ACCESS_READWRITE,  \
         .tlv.p = (tlv_array), \
         .info = snd_soc_info_volsw_2r, \
         .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
-        .private_value = (reg_left) | ((shift) << 8)  |         \
-                ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+        .private_value = (unsigned long)&(struct soc_mixer_control) \
+                {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+                .max = xmax, .invert = xinvert} }
 
 static const struct snd_kcontrol_new wm8580_snd_controls[] = {
 SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
         reg = wm8580_read(codec, WM8580_PLLA4 + offset);
         reg &= ~0x3f;
         reg |= pll_div.prescale | pll_div.postscale << 1 |
-                pll_div.freqmode << 4;
+                pll_div.freqmode << 3;
 
         wm8580_write(codec, WM8580_PLLA4 + offset, reg);
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3f..40cd274eb1ef 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
         inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
         &wm8990_dapm_ainlmux_controls, inmixer_event,
         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
         inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
         &wm8990_dapm_ainrmux_controls, inmixer_event,
         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"LIN12 PGA", "LIN2 Switch", "LIN2"},
         /* LIN34 PGA */
         {"LIN34 PGA", "LIN3 Switch", "LIN3"},
-        {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+        {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
         /* INMIXL */
         {"INMIXL", "Record Left Volume", "LOMIX"},
         {"INMIXL", "LIN2 Volume", "LIN2"},
         {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
         {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
-        /* AILNMUX */
-        {"AILNMUX", "INMIXL Mix", "INMIXL"},
-        {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
-        {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
-        {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
-        {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+        /* AINLMUX */
+        {"AINLMUX", "INMIXL Mix", "INMIXL"},
+        {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+        {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+        {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+        {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
         /* ADC */
-        {"Left ADC", NULL, "AILNMUX"},
+        {"Left ADC", NULL, "AINLMUX"},
 
         /* RIN12 PGA */
         {"RIN12 PGA", "RIN1 Switch", "RIN1"},
         {"RIN12 PGA", "RIN2 Switch", "RIN2"},
         /* RIN34 PGA */
         {"RIN34 PGA", "RIN3 Switch", "RIN3"},
-        {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+        {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
         /* INMIXL */
         {"INMIXR", "Record Right Volume", "ROMIX"},
         {"INMIXR", "RIN2 Volume", "RIN2"},
         {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
         {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
-        /* AIRNMUX */
-        {"AIRNMUX", "INMIXR Mix", "INMIXR"},
-        {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
-        {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
-        {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
-        {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+        /* AINRMUX */
+        {"AINRMUX", "INMIXR Mix", "INMIXR"},
+        {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+        {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+        {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+        {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
         /* ADC */
-        {"Right ADC", NULL, "AIRNMUX"},
+        {"Right ADC", NULL, "AINRMUX"},
 
         /* LOMIX */
         {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
 
         /* OUT3MIX */
-        {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+        {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
         {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
 
         /* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         /* Output Pins */
         {"LON", NULL, "LONMIX"},
         {"LOP", NULL, "LOPMIX"},
-        {"OUT", NULL, "OUT3MIX"},
+        {"OUT3", NULL, "OUT3MIX"},
         {"LOUT", NULL, "LOUT PGA"},
         {"SPKN", NULL, "SPKMIX"},
         {"ROUT", NULL, "ROUT PGA"},
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 6e23a81dba78..c2d1a7a18fa3 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm9705_soc_suspend(struct platform_device *pdev)
+static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg)
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_codec *codec = socdev->card->codec;
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657e..411a710be660 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
         tristate
 
 config SND_DAVINCI_SOC_EVM
-        tristate "SoC Audio support for DaVinci EVM"
-        depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+        tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+        depends on SND_DAVINCI_SOC
+        depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
         select SND_DAVINCI_SOC_I2S
         select SND_SOC_TLV320AIC3X
         help
           Say Y if you want to add support for SoC audio on TI
-          DaVinci EVM platform.
+          DaVinci DM6446 or DM355 EVM platforms.
 
 config SND_DAVINCI_SOC_SFFSDR
         tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007c..58fd1cbedd88 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
 
 /* evm audio private data */
 static struct aic3x_setup_data evm_aic3x_setup = {
-        .i2c_bus = 0,
+        .i2c_bus = 1,
         .i2c_address = 0x1b,
 };
 
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
         .codec_data = &evm_aic3x_setup,
 };
 
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
 static struct resource evm_snd_resources[] = {
         {
-                .start = DAVINCI_MCBSP_BASE,
-                .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+                .start = DAVINCI_ASP0_BASE,
+                .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
                 .flags = IORESOURCE_MEM,
         },
 };
 
 static struct evm_snd_platform_data evm_snd_data = {
-        .tx_dma_ch      = DM644X_DMACH_MCBSP_TX,
-        .rx_dma_ch      = DM644X_DMACH_MCBSP_RX,
+        .tx_dma_ch      = DAVINCI_DMA_ASP0_TX,
+        .rx_dma_ch      = DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+        {
+                .start = DAVINCI_ASP1_BASE,
+                .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+                .flags = IORESOURCE_MEM,
+        },
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+        .tx_dma_ch      = DAVINCI_DMA_ASP1_TX,
+        .rx_dma_ch      = DAVINCI_DMA_ASP1_RX,
 };
 
 static struct platform_device *evm_snd_device;
 
 static int __init evm_init(void)
 {
+        struct resource *resources;
+        unsigned num_resources;
+        struct evm_snd_platform_data *data;
+        int index;
         int ret;
 
-        evm_snd_device = platform_device_alloc("soc-audio", 0);
+        if (machine_is_davinci_evm()) {
+                davinci_cfg_reg(DM644X_MCBSP);
+
+                resources = evm_snd_resources;
+                num_resources = ARRAY_SIZE(evm_snd_resources);
+                data = &evm_snd_data;
+                index = 0;
+        } else if (machine_is_davinci_dm355_evm()) {
+                /* we don't use ASP1 IRQs, or we'd need to mux them ... */
+                davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+                davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+                resources = dm335evm_snd_resources;
+                num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+                data = &dm335evm_snd_data;
+                index = 1;
+        } else
+                return -EINVAL;
+
+        evm_snd_device = platform_device_alloc("soc-audio", index);
         if (!evm_snd_device)
                 return -ENOMEM;
 
         platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
         evm_snd_devdata.dev = &evm_snd_device->dev;
-        platform_device_add_data(evm_snd_device, &evm_snd_data,
-                                 sizeof(evm_snd_data));
+        platform_device_add_data(evm_snd_device, data, sizeof(*data));
 
-        ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
-                                            ARRAY_SIZE(evm_snd_resources));
+        ret = platform_device_add_resources(evm_snd_device, resources,
+                        num_resources);
         if (ret) {
                 platform_device_put(evm_snd_device);
                 return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d8..b1ea52fc83c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
 
 #include "davinci-pcm.h"
 
+
+/*
+ * NOTE:  terminology here is confusing.
+ *
+ *  - This driver supports the "Audio Serial Port" (ASP),
+ *    found on dm6446, dm355, and other DaVinci chips.
+ *
+ *  - But it labels it a "Multi-channel Buffered Serial Port"
+ *    (McBSP) as on older chips like the dm642 ... which was
+ *    backward-compatible, possibly explaining that confusion.
+ *
+ *  - OMAP chips have a controller called McBSP, which is
+ *    incompatible with the DaVinci flavor of McBSP.
+ *
+ *  - Newer DaVinci chips have a controller called McASP,
+ *    incompatible with ASP and with either McBSP.
+ *
+ * In short:  this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
 #define DAVINCI_MCBSP_DRR_REG   0x00
 #define DAVINCI_MCBSP_DXR_REG   0x04
 #define DAVINCI_MCBSP_SPCR_REG  0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_card *card = socdev->card;
-        struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+        struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
         struct davinci_mcbsp_dev *dev;
         struct resource *mem, *ioarea;
         struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 
         cpu_dai->private_data = dev;
 
-        dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+        dev->clk = clk_get(&pdev->dev, NULL);
         if (IS_ERR(dev->clk)) {
                 ret = -ENODEV;
                 goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_card *card = socdev->card;
-        struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+        struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
         struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
         struct resource *mem;
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53d..a05996588489 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
+#include <mach/edma.h>
 
 #include "davinci-pcm.h"
 
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
         spinlock_t lock;
         int period;             /* current DMA period */
         int master_lch;         /* Master DMA channel */
-        int slave_lch;          /* Slave DMA channel */
+        int slave_lch;          /* linked parameter RAM reload slot */
         struct davinci_pcm_dma_params *params;  /* DMA params */
 };
 
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
                 dst_bidx = data_type;
         }
 
-        davinci_set_dma_src_params(lch, src, INCR, W8BIT);
-        davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
-        davinci_set_dma_src_index(lch, src_bidx, 0);
-        davinci_set_dma_dest_index(lch, dst_bidx, 0);
-        davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+        edma_set_src(lch, src, INCR, W8BIT);
+        edma_set_dest(lch, dst, INCR, W8BIT);
+        edma_set_src_index(lch, src_bidx, 0);
+        edma_set_dest_index(lch, dst_bidx, 0);
+        edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
 
         prtd->period++;
         if (unlikely(prtd->period >= runtime->periods))
                 prtd->period = 0;
 }
 
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 {
         struct snd_pcm_substream *substream = data;
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
-        int tcc = TCC_ANY;
+        struct edmacc_param p_ram;
         int ret;
 
         if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
         prtd->params = dma_data;
 
         /* Request master DMA channel */
-        ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+        ret = edma_alloc_channel(prtd->params->channel,
                                   davinci_pcm_dma_irq, substream,
-                                  &prtd->master_lch, &tcc, EVENTQ_0);
-        if (ret)
+                                  EVENTQ_0);
+        if (ret < 0)
                 return ret;
+        prtd->master_lch = ret;
 
-        /* Request slave DMA channel */
-        ret = davinci_request_dma(PARAM_ANY, "Link",
-                                  NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
-        if (ret) {
-                davinci_free_dma(prtd->master_lch);
+        /* Request parameter RAM reload slot */
+        ret = edma_alloc_slot(EDMA_SLOT_ANY);
+        if (ret < 0) {
+                edma_free_channel(prtd->master_lch);
                 return ret;
         }
-
-        /* Link slave DMA channel in loopback */
-        davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+        prtd->slave_lch = ret;
+
+        /* Issue transfer completion IRQ when the channel completes a
+         * transfer, then always reload from the same slot (by a kind
+         * of loopback link).  The completion IRQ handler will update
+         * the reload slot with a new buffer.
+         *
+         * REVISIT save p_ram here after setting up everything except
+         * the buffer and its length (ccnt) ... use it as a template
+         * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+         */
+        edma_read_slot(prtd->slave_lch, &p_ram);
+        p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+        p_ram.link_bcntrld = prtd->slave_lch << 5;
+        edma_write_slot(prtd->slave_lch, &p_ram);
 
         return 0;
 }
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_RESUME:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-                davinci_start_dma(prtd->master_lch);
+                edma_start(prtd->master_lch);
                 break;
         case SNDRV_PCM_TRIGGER_STOP:
         case SNDRV_PCM_TRIGGER_SUSPEND:
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-                davinci_stop_dma(prtd->master_lch);
+                edma_stop(prtd->master_lch);
                 break;
         default:
                 ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
 {
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
-        struct paramentry_descriptor temp;
+        struct edmacc_param temp;
 
         prtd->period = 0;
         davinci_pcm_enqueue_dma(substream);
 
-        /* Get slave channel dma params for master channel startup */
-        davinci_get_dma_params(prtd->slave_lch, &temp);
-        davinci_set_dma_params(prtd->master_lch, &temp);
+        /* Copy self-linked parameter RAM entry into master channel */
+        edma_read_slot(prtd->slave_lch, &temp);
+        edma_write_slot(prtd->master_lch, &temp);
 
         return 0;
 }
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
 
         spin_lock(&prtd->lock);
 
-        davinci_dma_getposition(prtd->master_lch, &src, &dst);
+        edma_get_position(prtd->master_lch, &src, &dst);
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                 count = src - runtime->dma_addr;
         else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct davinci_runtime_data *prtd = runtime->private_data;
 
-        davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+        edma_unlink(prtd->slave_lch);
 
-        davinci_free_dma(prtd->slave_lch);
-        davinci_free_dma(prtd->master_lch);
+        edma_free_slot(prtd->slave_lch);
+        edma_free_channel(prtd->master_lch);
 
         kfree(prtd);
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 
 static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
         .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_BATCH,
         .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
                    SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
         .rate_min = 8000,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a6d1178ce128..91ef17992de5 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void)
 module_init(n810_soc_init);
 module_exit(n810_soc_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("ALSA SoC Nokia N810");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 90f4df7fd906..912614283848 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +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
@@ -532,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void)
 }
 module_exit(snd_omap_mcbsp_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index df7ad13ba73d..c8147aace813 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +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
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 1bdbb0427183..07cf7f46b584 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +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
@@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void)
 }
 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 --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index e4369bdfd77d..8d9d26916b05 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +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
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 48a73f64500b..44fcc4e01e08 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = {
 
 static struct platform_device *palm27x_snd_device;
 
-static int __init palm27x_asoc_init(void)
+static int palm27x_asoc_probe(struct platform_device *pdev)
 {
         int ret;
 
@@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void)
                 machine_is_palmld()))
                 return -ENODEV;
 
+        if (pdev->dev.platform_data)
+                palm27x_ep_gpio = ((struct palm27x_asoc_info *)
+                        (pdev->dev.platform_data))->jack_gpio;
+
         ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
         if (ret)
                 return ret;
@@ -245,16 +249,31 @@ err_alloc:
         return ret;
 }
 
-static void __exit palm27x_asoc_exit(void)
+static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
 {
         free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
         gpio_free(palm27x_ep_gpio);
         platform_device_unregister(palm27x_snd_device);
+        return 0;
 }
 
-void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+static struct platform_driver palm27x_wm9712_driver = {
+        .probe          = palm27x_asoc_probe,
+        .remove         = __devexit_p(palm27x_asoc_remove),
+        .driver         = {
+                .name           = "palm27x-asoc",
+                .owner          = THIS_MODULE,
+        },
+};
+
+static int __init palm27x_asoc_init(void)
+{
+        return platform_driver_register(&palm27x_wm9712_driver);
+}
+
+static void __exit palm27x_asoc_exit(void)
 {
-        palm27x_ep_gpio = data->jack_gpio;
+        platform_driver_unregister(&palm27x_wm9712_driver);
 }
 
 module_init(palm27x_asoc_init);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index de2254475d52..286be31545df 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
  * 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 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
                 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 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
                 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 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
         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.
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 689ffcd17e1f..ab680aac3fcb 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 
         return snd_soc_register_dai(dai);
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
         .info = (SNDRV_PCM_INFO_MMAP |
                 SNDRV_PCM_INFO_INTERLEAVED |
                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                SNDRV_PCM_INFO_MMAP_VALID),
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_BATCH),
         .formats =      DMABRG_FMTS,
         .rates =        DMABRG_RATES,
         .rate_min =             8000,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0d..1cd149b9ce69 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev)
         struct snd_soc_platform *platform = card->platform;
         struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
+        if (!card->instantiated)
+                return 0;
+
         run_delayed_work(&card->delayed_work);
 
         if (platform->remove)