Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.7

Nothing too exciting except for the ams-delta change which is relatively
lerge due to the fact that the driver loading had been totally broken as
the driver needed a newer API to function.

11 files changed, 78 insertions, 62 deletions

diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index bcbf1d00aa85..99f32f7c0692 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,8 +1,9 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
 snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o
 
-snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o
-obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
+ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),)
+snd-soc-core-objs += soc-dmaengine-pcm.o
+endif
 
 obj-$(CONFIG_SND_SOC)   += snd-soc-core.o
 obj-$(CONFIG_SND_SOC)   += codecs/
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 185d8dd36399..f379b085c392 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -178,6 +178,12 @@
 #define DA9055_AIF_WORD_S24_LE          (2 << 2)
 #define DA9055_AIF_WORD_S32_LE          (3 << 2)
 
+/* MIC_L_CTRL bit fields */
+#define DA9055_MIC_L_MUTE_EN            (1 << 6)
+
+/* MIC_R_CTRL bit fields */
+#define DA9055_MIC_R_MUTE_EN            (1 << 6)
+
 /* MIXIN_L_CTRL bit fields */
 #define DA9055_MIXIN_L_MIX_EN           (1 << 3)
 
@@ -476,7 +482,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
                              struct snd_ctl_elem_value *ucontrol)
 {
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-        u8 reg_val, adc_left, adc_right;
+        u8 reg_val, adc_left, adc_right, mic_left, mic_right;
         int avg_left_data, avg_right_data, offset_l, offset_r;
 
         if (ucontrol->value.integer.value[0]) {
@@ -485,6 +491,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
                  * offsets must be done first
                  */
 
+                /* Save current values from Mic control registers */
+                mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL);
+                mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL);
+
+                /* Mute Mic PGA Left and Right */
+                snd_soc_update_bits(codec, DA9055_MIC_L_CTRL,
+                                    DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN);
+                snd_soc_update_bits(codec, DA9055_MIC_R_CTRL,
+                                    DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN);
+
                 /* Save current values from ADC control registers */
                 adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL);
                 adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL);
@@ -520,6 +536,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol,
                 /* Restore original values of ADC control registers */
                 snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left);
                 snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right);
+
+                /* Restore original values of Mic control registers */
+                snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left);
+                snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right);
         }
 
         return snd_soc_put_volsw(kcontrol, ucontrol);
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index e8f97af75928..00b85cc1b9a3 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = {
         {"VIBRA DAC", NULL, "Vibra Playback"},
 
         /* ADC -> Stream mapping */
-        {"ADC Left", NULL, "Legacy Capture"},
-        {"ADC Left", NULL, "Capture"},
-        {"ADC Right", NULL, "Legacy Capture"},
-        {"ADC Right", NULL, "Capture"},
+        {"Legacy Capture" , NULL, "ADC Left"},
+        {"Capture", NULL, "ADC Left"},
+        {"Legacy Capture", NULL, "ADC Right"},
+        {"Capture" , NULL, "ADC Right"},
 
         /* Capture path */
         {"Analog Left Capture Route", "Headset Mic", "HSMIC"},
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index efa93dbb0191..eab64a193989 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
                  WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0,
                  digital_tlv),
 SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
-           WM2200_SPK1R_MUTE_SHIFT, 1, 0),
+           WM2200_SPK1R_MUTE_SHIFT, 1, 1),
 };
 
 WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
@@ -2091,6 +2091,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c,
 
         switch (wm2200->rev) {
         case 0:
+        case 1:
                 ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch,
                                             ARRAY_SIZE(wm2200_reva_patch));
                 if (ret != 0) {
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index dc0ee7626626..d8e96b2cd03e 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -575,56 +575,53 @@ static struct snd_soc_card ams_delta_audio_card = {
 };
 
 /* Module init/exit */
-static struct platform_device *ams_delta_audio_platform_device;
-static struct platform_device *cx20442_platform_device;
-
-static int __init ams_delta_module_init(void)
+static __devinit int ams_delta_probe(struct platform_device *pdev)
 {
+        struct snd_soc_card *card = &ams_delta_audio_card;
         int ret;
 
-        if (!(machine_is_ams_delta()))
-                return -ENODEV;
-
-        ams_delta_audio_platform_device =
-                        platform_device_alloc("soc-audio", -1);
-        if (!ams_delta_audio_platform_device)
-                return -ENOMEM;
+        card->dev = &pdev->dev;
 
-        platform_set_drvdata(ams_delta_audio_platform_device,
-                                &ams_delta_audio_card);
-
-        ret = platform_device_add(ams_delta_audio_platform_device);
-        if (ret)
-                goto err;
-
-        /*
-         * Codec platform device could be registered from elsewhere (board?),
-         * but I do it here as it makes sense only if used with the card.
-         */
-        cx20442_platform_device =
-                platform_device_register_simple("cx20442-codec", -1, NULL, 0);
+        ret = snd_soc_register_card(card);
+        if (ret) {
+                dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+                card->dev = NULL;
+                return ret;
+        }
         return 0;
-err:
-        platform_device_put(ams_delta_audio_platform_device);
-        return ret;
 }
-late_initcall(ams_delta_module_init);
 
-static void __exit ams_delta_module_exit(void)
+static int __devexit ams_delta_remove(struct platform_device *pdev)
 {
+        struct snd_soc_card *card = platform_get_drvdata(pdev);
+
         if (tty_unregister_ldisc(N_V253) != 0)
-                dev_warn(&ams_delta_audio_platform_device->dev,
+                dev_warn(&pdev->dev,
                         "failed to unregister V253 line discipline\n");
 
         snd_soc_jack_free_gpios(&ams_delta_hook_switch,
                         ARRAY_SIZE(ams_delta_hook_switch_gpios),
                         ams_delta_hook_switch_gpios);
 
-        platform_device_unregister(cx20442_platform_device);
-        platform_device_unregister(ams_delta_audio_platform_device);
+        snd_soc_unregister_card(card);
+        card->dev = NULL;
+        return 0;
 }
-module_exit(ams_delta_module_exit);
+
+#define DRV_NAME "ams-delta-audio"
+
+static struct platform_driver ams_delta_driver = {
+        .driver = {
+                .name = DRV_NAME,
+                .owner = THIS_MODULE,
+        },
+        .probe = ams_delta_probe,
+        .remove = __devexit_p(ams_delta_remove),
+};
+
+module_platform_driver(ams_delta_driver);
 
 MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
 MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 4a73ef3ae12f..a57a4e68dcc6 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -216,7 +216,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
         twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
         twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
         twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
-        twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+        twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator");
         twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
         twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
         twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index c02b001ee4b5..56965bb3275c 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -40,7 +40,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <plat/omap_hwmod.h>
 #include "omap-mcpdm.h"
 #include "omap-pcm.h"
 
@@ -260,13 +259,9 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
         mutex_lock(&mcpdm->mutex);
 
         if (!dai->active) {
-                /* Enable watch dog for ES above ES 1.0 to avoid saturation */
-                if (omap_rev() != OMAP4430_REV_ES1_0) {
-                        u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+                u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
 
-                        omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL,
-                                         ctrl | MCPDM_WD_EN);
-                }
+                omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN);
                 omap_mcpdm_open_streams(mcpdm);
         }
         mutex_unlock(&mcpdm->mutex);
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
index 73ac5463c9e4..e834faf859fd 100644
--- a/sound/soc/pxa/mmp-pcm.c
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -15,13 +15,13 @@
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
 #include <linux/dmaengine.h>
+#include <linux/platform_data/dma-mmp_tdma.h>
 #include <linux/platform_data/mmp_audio.h>
 #include <sound/pxa2xx-lib.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
-#include <mach/sram.h>
 #include <sound/dmaengine_pcm.h>
 
 struct mmp_dma_data {
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
index 5dc10dfc0d42..b0d46d63d55e 100644
--- a/sound/soc/samsung/bells.c
+++ b/sound/soc/samsung/bells.c
@@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = {
         {
                 .name = "Sub",
                 .stream_name = "Sub",
-                .cpu_dai_name = "wm5102-aif3",
+                .cpu_dai_name = "wm5110-aif3",
                 .codec_dai_name = "wm9081-hifi",
                 .codec_name = "wm9081.1-006c",
                 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
@@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = {
         {
                 .name = "Sub",
                 .stream_name = "Sub",
-                .cpu_dai_name = "wm5102-aif3",
+                .cpu_dai_name = "wm5110-aif3",
                 .codec_dai_name = "wm9081-hifi",
                 .codec_name = "wm9081.1-006c",
                 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 5328ae5539f1..9d7f30774a44 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -20,6 +20,7 @@
 #include <linux/sh_dma.h>
 #include <linux/slab.h>
 #include <linux/module.h>
+#include <linux/workqueue.h>
 #include <sound/soc.h>
 #include <sound/sh_fsi.h>
 
@@ -223,7 +224,7 @@ struct fsi_stream {
          */
         struct dma_chan         *chan;
         struct sh_dmae_slave    slave; /* see fsi_handler_init() */
-        struct tasklet_struct   tasklet;
+        struct work_struct      work;
         dma_addr_t              dma;
 };
 
@@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data)
         snd_pcm_period_elapsed(io->substream);
 }
 
-static void fsi_dma_do_tasklet(unsigned long data)
+static void fsi_dma_do_work(struct work_struct *work)
 {
-        struct fsi_stream *io = (struct fsi_stream *)data;
+        struct fsi_stream *io = container_of(work, struct fsi_stream, work);
         struct fsi_priv *fsi = fsi_stream_to_priv(io);
         struct snd_soc_dai *dai;
         struct dma_async_tx_descriptor *desc;
@@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data)
          * FIXME
          *
          * In DMAEngine case, codec and FSI cannot be started simultaneously
-         * since FSI is using tasklet.
+         * since FSI is using the scheduler work queue.
          * Therefore, in capture case, probably FSI FIFO will have got
          * overflow error in this point.
          * in that case, DMA cannot start transfer until error was cleared.
@@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param)
 
 static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io)
 {
-        tasklet_schedule(&io->tasklet);
+        schedule_work(&io->work);
 
         return 0;
 }
@@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev
                 return fsi_stream_probe(fsi, dev);
         }
 
-        tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
+        INIT_WORK(&io->work, fsi_dma_do_work);
 
         return 0;
 }
 
 static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io)
 {
-        tasklet_kill(&io->tasklet);
+        cancel_work_sync(&io->work);
 
         fsi_stream_stop(fsi, io);
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index fa0fd8ddae90..1ab5fe04bfcc 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -22,7 +22,7 @@
 
 /**
  * snd_soc_jack_new - Create a new jack
- * @card:  ASoC card
+ * @codec: ASoC codec
  * @id:    an identifying string for this jack
  * @type:  a bitmask of enum snd_jack_type values that can be detected by
  *         this jack
@@ -133,12 +133,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones);
 
 /**
  * snd_soc_jack_get_type - Based on the mic bias value, this function returns
- * the type of jack from the zones delcared in the jack type
+ * the type of jack from the zones declared in the jack type
  *
+ * @jack:  ASoC jack
  * @micbias_voltage:  mic bias voltage at adc channel when jack is plugged in
  *
  * Based on the mic bias value passed, this function helps identify
- * the type of jack from the already delcared jack zones
+ * the type of jack from the already declared jack zones
  */
 int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage)
 {