16 files changed, 51 insertions, 87 deletions

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index f8e10ced244..b3e24f28942 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -140,7 +140,7 @@
  * min : 0xFE : -115.0 dB
  * mute: 0xFF
  */
-static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
 
 static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d1926266fe0..8e92fb88ed0 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 }
 
 /*
- * using codec assist to small pop, hp_powerup or lineout_powerup
- * should stay setting until vag_powerup is fully ramped down,
- * vag fully ramped down require 400ms.
+ * As manual described, ADC/DAC only works when VAG powerup,
+ * So enabled VAG before ADC/DAC up.
+ * In power down case, we need wait 400ms when vag fully ramped down.
  */
-static int small_pop_event(struct snd_soc_dapm_widget *w,
+static int power_vag_event(struct snd_soc_dapm_widget *w,
         struct snd_kcontrol *kcontrol, int event)
 {
         switch (event) {
@@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w,
                         SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
                 break;
 
-        case SND_SOC_DAPM_PRE_PMD:
+        case SND_SOC_DAPM_POST_PMD:
                 snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
                         SGTL5000_VAG_POWERUP, 0);
                 msleep(400);
@@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
                                 mic_bias_event,
                                 SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-        SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
-                        small_pop_event,
-                        SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
-        SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0,
-                        small_pop_event,
-                        SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+        SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
 
         SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
         SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux),
@@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
                                 0, SGTL5000_CHIP_DIG_POWER,
                                 1, 0),
 
-        SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
+        SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0,
+                            power_vag_event,
+                            SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
+        SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
         SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
 };
 
@@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
         {"Capture Mux", "LINE_IN", "LINE_IN"},  /* line_in --> adc_mux */
         {"Capture Mux", "MIC_IN", "MIC_IN"},    /* mic_in --> adc_mux */
 
+        {"ADC", NULL, "VAG_POWER"},
         {"ADC", NULL, "Capture Mux"},           /* adc_mux --> adc */
         {"AIFOUT", NULL, "ADC"},                /* adc --> i2s_out */
 
+        {"DAC", NULL, "VAG_POWER"},
         {"DAC", NULL, "AIFIN"},                 /* i2s-->dac,skip audio mux */
         {"Headphone Mux", "DAC", "DAC"},        /* dac --> hp_mux */
         {"LO", NULL, "DAC"},                    /* dac --> line_out */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 10d27890eed..7c49642af05 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2181,26 +2181,9 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
         case SND_SOC_BIAS_STANDBY:
                 if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
                         switch (control->type) {
-                        case WM8994:
-                                if (wm8994->revision < 4) {
-                                        /* Tweak DC servo and DSP
-                                         * configuration for improved
-                                         * performance. */
-                                        snd_soc_write(codec, 0x102, 0x3);
-                                        snd_soc_write(codec, 0x56, 0x3);
-                                        snd_soc_write(codec, 0x817, 0);
-                                        snd_soc_write(codec, 0x102, 0);
-                                }
-                                break;
-
                         case WM8958:
                                 if (wm8994->revision == 0) {
                                         /* Optimise performance for rev A */
-                                        snd_soc_write(codec, 0x102, 0x3);
-                                        snd_soc_write(codec, 0xcb, 0x81);
-                                        snd_soc_write(codec, 0x817, 0);
-                                        snd_soc_write(codec, 0x102, 0);
-
                                         snd_soc_update_bits(codec,
                                                             WM8958_CHARGE_PUMP_2,
                                                             WM8958_CP_DISCH,
@@ -2208,13 +2191,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
                                 }
                                 break;
 
-                        case WM1811:
-                                if (wm8994->revision < 2) {
-                                        snd_soc_write(codec, 0x102, 0x3);
-                                        snd_soc_write(codec, 0x5d, 0x7e);
-                                        snd_soc_write(codec, 0x5e, 0x0);
-                                        snd_soc_write(codec, 0x102, 0x0);
-                                }
+                        default:
                                 break;
                         }
 
diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c
index a839494c5ea..0fe66c3dde1 100644
--- a/sound/soc/imx/imx-audmux.c
+++ b/sound/soc/imx/imx-audmux.c
@@ -79,14 +79,17 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
         if (!buf)
                 return -ENOMEM;
 
+        if (!audmux_base)
+                return -ENOSYS;
+
         if (audmux_clk)
-                clk_enable(audmux_clk);
+                clk_prepare_enable(audmux_clk);
 
         ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port));
         pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port));
 
         if (audmux_clk)
-                clk_disable(audmux_clk);
+                clk_disable_unprepare(audmux_clk);
 
         ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
                        pdcr, ptcr);
@@ -158,7 +161,7 @@ static void __init audmux_debugfs_init(void)
                 return;
         }
 
-        for (i = 1; i < 8; i++) {
+        for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) {
                 snprintf(buf, sizeof(buf), "ssi%d", i);
                 if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
                                          (void *)i, &audmux_debugfs_fops))
@@ -237,13 +240,13 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr,
                 return -ENOSYS;
 
         if (audmux_clk)
-                clk_enable(audmux_clk);
+                clk_prepare_enable(audmux_clk);
 
         writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port));
         writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port));
 
         if (audmux_clk)
-                clk_disable(audmux_clk);
+                clk_disable_unprepare(audmux_clk);
 
         return 0;
 }
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index e43c8fa2788..6b818de2fc0 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -21,6 +21,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dmaengine.h>
+#include <linux/types.h>
 
 #include <sound/core.h>
 #include <sound/initval.h>
@@ -58,6 +59,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
         if (ret)
                 return ret;
 
+        slave_config.device_fc = false;
+
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
                 slave_config.dst_addr = dma_params->dma_addr;
                 slave_config.dst_maxburst = dma_params->burstsize;
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 6ca1f46d84a..e373fbbc97a 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -28,6 +28,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dmaengine.h>
+#include <linux/fsl/mxs-dma.h>
 
 #include <sound/core.h>
 #include <sound/initval.h>
@@ -36,7 +37,6 @@
 #include <sound/soc.h>
 #include <sound/dmaengine_pcm.h>
 
-#include <mach/dma.h>
 #include "mxs-pcm.h"
 
 struct mxs_pcm_dma_data {
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index 12be05b1688..53f4fd8fece 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -24,12 +24,12 @@
 #include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/time.h>
+#include <linux/fsl/mxs-dma.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/saif.h>
-#include <mach/dma.h>
 #include <asm/mach-types.h>
 #include <mach/hardware.h>
 #include <mach/mxs.h>
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 49fe63ce51f..7d4fa8ed669 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = {
 };
 
 
-/* Board specific codec bias level control */
-static int ams_delta_set_bias_level(struct snd_soc_card *card,
-                                    struct snd_soc_dapm_context *dapm,
-                                    enum snd_soc_bias_level level)
-{
-        switch (level) {
-        case SND_SOC_BIAS_ON:
-        case SND_SOC_BIAS_PREPARE:
-        case SND_SOC_BIAS_STANDBY:
-                if (card->dapm.bias_level == SND_SOC_BIAS_OFF)
-                        ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
-                                                AMS_DELTA_LATCH2_MODEM_NRESET);
-                break;
-        case SND_SOC_BIAS_OFF:
-                if (card->dapm.bias_level != SND_SOC_BIAS_OFF)
-                        ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
-                                                0);
-        }
-        card->dapm.bias_level = level;
-
-        return 0;
-}
-
 /* Digital mute implemented using modem/CPU multiplexer.
  * Shares hardware with codec config pulse generation */
 static bool ams_delta_muted = 1;
@@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
                 ams_delta_ops.shutdown = ams_delta_shutdown;
         }
 
-        /* Set codec bias level */
-        ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
-
         /* Add hook switch - can be used to control the codec from userspace
          * even if line discipline fails */
         ret = snd_soc_jack_new(rtd->codec, "hook_switch",
@@ -598,7 +572,6 @@ static struct snd_soc_card ams_delta_audio_card = {
         .owner = THIS_MODULE,
         .dai_link = &ams_delta_dai_link,
         .num_links = 1,
-        .set_bias_level = ams_delta_set_bias_level,
 };
 
 /* Module init/exit */
@@ -635,7 +608,7 @@ err:
         platform_device_put(ams_delta_audio_platform_device);
         return ret;
 }
-module_init(ams_delta_module_init);
+late_initcall(ams_delta_module_init);
 
 static void __exit ams_delta_module_exit(void)
 {
@@ -647,11 +620,6 @@ static void __exit ams_delta_module_exit(void)
                         ARRAY_SIZE(ams_delta_hook_switch_gpios),
                         ams_delta_hook_switch_gpios);
 
-        /* Keep modem power on */
-        ams_delta_set_bias_level(&ams_delta_audio_card,
-                                 &ams_delta_audio_card.rtd[0].codec->dapm,
-                                 SND_SOC_BIAS_STANDBY);
-
         platform_device_unregister(cx20442_platform_device);
         platform_device_unregister(ams_delta_audio_platform_device);
 }
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 4800d5fe568..06ea2744cc8 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -11,6 +11,7 @@
  */
 
 #include <linux/init.h>
+#include <linux/io.h>
 #include <linux/module.h>
 #include <linux/platform_device.h>
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 609abd51e55..d08583790d2 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -17,6 +17,7 @@
 #include <linux/delay.h>
 #include <linux/clk.h>
 #include <linux/platform_device.h>
+#include <linux/io.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index f3417f2311b..fe3995ce9b3 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,8 +1,8 @@
 config SND_SOC_SAMSUNG
         tristate "ASoC support for Samsung"
-        depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4
+        depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4
         select S3C64XX_DMA if ARCH_S3C64XX
-        select S3C2410_DMA if ARCH_S3C2410
+        select S3C2410_DMA if ARCH_S3C24XX
         help
           Say Y or M if you want to add support for codecs attached to
           the Samsung SoCs' Audio interfaces. You will also need to
@@ -84,7 +84,7 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710
 
 config SND_SOC_SAMSUNG_LN2440SBC_ALC650
         tristate "SoC AC97 Audio support for LN2440SBC - ALC650"
-        depends on SND_SOC_SAMSUNG && ARCH_S3C2410
+        depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
         select S3C2410_DMA
         select AC97_BUS
         select SND_SOC_AC97_CODEC
@@ -95,7 +95,7 @@ config SND_SOC_SAMSUNG_LN2440SBC_ALC650
 
 config SND_SOC_SAMSUNG_S3C24XX_UDA134X
         tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
-        depends on SND_SOC_SAMSUNG && ARCH_S3C2410
+        depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
         select SND_S3C24XX_I2S
         select SND_SOC_L3
         select SND_SOC_UDA134X
@@ -107,14 +107,14 @@ config SND_SOC_SAMSUNG_SIMTEC
 
 config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23
         tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
-        depends on SND_SOC_SAMSUNG && ARCH_S3C2410
+        depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
         select SND_S3C24XX_I2S
         select SND_SOC_TLV320AIC23
         select SND_SOC_SAMSUNG_SIMTEC
 
 config SND_SOC_SAMSUNG_SIMTEC_HERMES
         tristate "SoC I2S Audio support for Simtec Hermes board"
-        depends on SND_SOC_SAMSUNG && ARCH_S3C2410
+        depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
         select SND_S3C24XX_I2S
         select SND_SOC_TLV320AIC3X
         select SND_SOC_SAMSUNG_SIMTEC
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 0193e595d41..5cfcc655e95 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -130,7 +130,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info,
         sg_dma_len(&sg) = size;
         sg_dma_address(&sg) = buff;
 
-        desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan,
+        desc = dmaengine_prep_slave_sg(siu_stream->chan,
                 &sg, 1, DMA_MEM_TO_DEV, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
         if (!desc) {
                 dev_err(dev, "Failed to allocate a dma descriptor\n");
@@ -180,7 +180,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info,
         sg_dma_len(&sg) = size;
         sg_dma_address(&sg) = buff;
 
-        desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan,
+        desc = dmaengine_prep_slave_sg(siu_stream->chan,
                 &sg, 1, DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
         if (!desc) {
                 dev_err(dev, "Failed to allocate dma descriptor\n");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a4deebc0801..8d2ebf502df 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1087,6 +1087,8 @@ static int soc_probe_platform(struct snd_soc_card *card,
                 snd_soc_dapm_new_controls(&platform->dapm,
                         driver->dapm_widgets, driver->num_dapm_widgets);
 
+        platform->dapm.idle_bias_off = 1;
+
         if (driver->probe) {
                 ret = driver->probe(platform);
                 if (ret < 0) {
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 33509de5254..e53349912b2 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused)
         struct tegra_i2s *i2s = s->private;
         int i;
 
+        clk_enable(i2s->clk_i2s);
+
         for (i = 0; i < ARRAY_SIZE(regs); i++) {
                 u32 val = tegra_i2s_read(i2s, regs[i].offset);
                 seq_printf(s, "%s = %08x\n", regs[i].name, val);
         }
 
+        clk_disable(i2s->clk_i2s);
+
         return 0;
 }
 
@@ -112,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
                 debugfs_remove(i2s->debug);
 }
 #else
-static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
+static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
 {
 }
 
diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
index 475428cf270..9ff2c601445 100644
--- a/sound/soc/tegra/tegra_spdif.c
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused)
         struct tegra_spdif *spdif = s->private;
         int i;
 
+        clk_enable(spdif->clk_spdif_out);
+
         for (i = 0; i < ARRAY_SIZE(regs); i++) {
                 u32 val = tegra_spdif_read(spdif, regs[i].offset);
                 seq_printf(s, "%s = %08x\n", regs[i].name, val);
         }
 
+        clk_disable(spdif->clk_spdif_out);
+
         return 0;
 }
 
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 21554611557..b609d2c64c5 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -132,7 +132,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
         sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)),
                     dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1));
         sg_dma_address(&sg) = buf_dma_addr;
-        desc = chan->device->device_prep_slave_sg(chan, &sg, 1,
+        desc = dmaengine_prep_slave_sg(chan, &sg, 1,
                 dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
                 DMA_MEM_TO_DEV : DMA_DEV_TO_MEM,
                 DMA_PREP_INTERRUPT | DMA_CTRL_ACK);