Merge tag 'sound-3.17-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This time it contains a bunch of small ASoC fixes that slipped from in
  previous updates, in addition to the usual HD-audio fixes and the
  regression fixes for FireWire updates in 3.17.

  All commits are reasonably small fixes"

* tag 'sound-3.17-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix COEF setups for ALC1150 codec
  ASoC: simple-card: Fix bug of wrong decrement DT node's refcount
  ALSA: hda - Fix digital mic on Acer Aspire 3830TG
  ASoC: omap-twl4030: Fix typo in 2nd dai link's platform_name
  ALSA: firewire-lib/dice: add arrangements of PCM pointer and interrupts for Dice quirk
  ALSA: dice: fix wrong channel mappping at higher sampling rate
  ASoC: cs4265: Fix setting of functional mode and clock divider
  ASoC: cs4265: Fix clock rates in clock map table
  ASoC: rt5677: correct mismatch widget name
  ASoC: rt5640: Do not allow regmap to use bulk read-write operations
  ASoC: tegra: Fix typo in include guard
  ASoC: da732x: Fix typo in include guard
  ASoC: core: fix .info for SND_SOC_BYTES_TLV
  ASoC: rcar: Use && instead of & for boolean expressions
  ASoC: Use dev_set_name() instead of init_name
  ASoC: axi: Fix ADI AXI SPDIF specification

16 files changed, 67 insertions, 28 deletions

diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
index 46f344965313..4eb7997674a0 100644
--- a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
+++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
@@ -1,7 +1,7 @@
 ADI AXI-SPDIF controller
 
 Required properties:
- - compatible : Must be "adi,axi-spdif-1.00.a"
+ - compatible : Must be "adi,axi-spdif-tx-1.00.a"
  - reg : Must contain SPDIF core's registers location and length
  - clocks : Pairs of phandle and specifier referencing the controller's clocks.
    The controller expects two clocks, the clock used for the AXI interface and
diff --git a/include/sound/soc.h b/include/sound/soc.h
index be6ecae247b0..c83a334dd00f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -277,7 +277,7 @@
         .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \
                   SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
         .tlv.c = (snd_soc_bytes_tlv_callback), \
-        .info = snd_soc_info_bytes_ext, \
+        .info = snd_soc_bytes_info_ext, \
         .private_value = (unsigned long)&(struct soc_bytes_ext) \
                 {.max = xcount, .get = xhandler_get, .put = xhandler_put, } }
 #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index f96bf4c7c232..95fc2eaf11dc 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s,
 static void update_pcm_pointers(struct amdtp_stream *s,
                                 struct snd_pcm_substream *pcm,
                                 unsigned int frames)
-{       unsigned int ptr;
+{
+        unsigned int ptr;
+
+        /*
+         * In IEC 61883-6, one data block represents one event. In ALSA, one
+         * event equals to one PCM frame. But Dice has a quirk to transfer
+         * two PCM frames in one data block.
+         */
+        if (s->double_pcm_frames)
+                frames *= 2;
 
         ptr = s->pcm_buffer_pointer + frames;
         if (ptr >= pcm->runtime->buffer_size)
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index d8ee7b0e9386..4823c08196ac 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -125,6 +125,7 @@ struct amdtp_stream {
         unsigned int pcm_buffer_pointer;
         unsigned int pcm_period_pointer;
         bool pointer_flush;
+        bool double_pcm_frames;
 
         struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
 
diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c
index a9a30c0161f1..e3a04d69c853 100644
--- a/sound/firewire/dice.c
+++ b/sound/firewire/dice.c
@@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream,
                 return err;
 
         /*
-         * At rates above 96 kHz, pretend that the stream runs at half the
-         * actual sample rate with twice the number of channels; two samples
-         * of a channel are stored consecutively in the packet. Requires
-         * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL.
+         * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in
+         * one data block of AMDTP packet. Thus sampling transfer frequency is
+         * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are
+         * transferred on AMDTP packets at 96 kHz. Two successive samples of a
+         * channel are stored consecutively in the packet. This quirk is called
+         * as 'Dual Wire'.
+         * For this quirk, blocking mode is required and PCM buffer size should
+         * be aligned to SYT_INTERVAL.
          */
         channels = params_channels(hw_params);
         if (rate_index > 4) {
@@ -579,18 +583,25 @@ static int dice_hw_params(struct snd_pcm_substream *substream,
                         return err;
                 }
 
-                for (i = 0; i < channels; i++) {
-                        dice->stream.pcm_positions[i * 2] = i;
-                        dice->stream.pcm_positions[i * 2 + 1] = i + channels;
-                }
-
                 rate /= 2;
                 channels *= 2;
+                dice->stream.double_pcm_frames = true;
+        } else {
+                dice->stream.double_pcm_frames = false;
         }
 
         mode = rate_index_to_mode(rate_index);
         amdtp_stream_set_parameters(&dice->stream, rate, channels,
                                     dice->rx_midi_ports[mode]);
+        if (rate_index > 4) {
+                channels /= 2;
+
+                for (i = 0; i < channels; i++) {
+                        dice->stream.pcm_positions[i] = i * 2;
+                        dice->stream.pcm_positions[i + channels] = i * 2 + 1;
+                }
+        }
+
         amdtp_stream_set_pcm_format(&dice->stream,
                                     params_format(hw_params));
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6f2fa838b635..6e5d0cb4e3d7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -217,6 +217,7 @@ enum {
         CXT_FIXUP_HEADPHONE_MIC_PIN,
         CXT_FIXUP_HEADPHONE_MIC,
         CXT_FIXUP_GPIO1,
+        CXT_FIXUP_ASPIRE_DMIC,
         CXT_FIXUP_THINKPAD_ACPI,
         CXT_FIXUP_OLPC_XO,
         CXT_FIXUP_CAP_MIX_AMP,
@@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = {
                         { }
                 },
         },
+        [CXT_FIXUP_ASPIRE_DMIC] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = cxt_fixup_stereo_dmic,
+                .chained = true,
+                .chain_id = CXT_FIXUP_GPIO1,
+        },
         [CXT_FIXUP_THINKPAD_ACPI] = {
                 .type = HDA_FIXUP_FUNC,
                 .v.func = hda_fixup_thinkpad_acpi,
@@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
         SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
-        SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1),
+        SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC),
         SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
         SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
         SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d446ac3137b3..1ba22fb527c2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -328,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
                 case 0x10ec0885:
                 case 0x10ec0887:
                 /*case 0x10ec0889:*/ /* this causes an SPDIF problem */
+                case 0x10ec0900:
                         alc889_coef_init(codec);
                         break;
                 case 0x10ec0888:
@@ -2350,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec)
         switch (codec->vendor_id) {
         case 0x10ec0882:
         case 0x10ec0885:
+        case 0x10ec0900:
                 break;
         default:
                 /* ALC883 and variants */
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30ca52c0..98523209f739 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = {
 
         /*64k*/
         {8192000, 64000, 1, 0},
-        {1228800, 64000, 1, 1},
-        {1693440, 64000, 1, 2},
-        {2457600, 64000, 1, 3},
-        {3276800, 64000, 1, 4},
+        {12288000, 64000, 1, 1},
+        {16934400, 64000, 1, 2},
+        {24576000, 64000, 1, 3},
+        {32768000, 64000, 1, 4},
 
         /* 88.2k */
         {11289600, 88200, 1, 0},
@@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
         index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
         if (index >= 0) {
                 snd_soc_update_bits(codec, CS4265_ADC_CTL,
-                        CS4265_ADC_FM, clk_map_table[index].fm_mode);
+                        CS4265_ADC_FM, clk_map_table[index].fm_mode << 6);
                 snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
                         CS4265_MCLK_FREQ_MASK,
-                        clk_map_table[index].mclkdiv);
+                        clk_map_table[index].mclkdiv << 4);
 
         } else {
                 dev_err(codec->dev, "can't get correct mclk\n");
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index 1dceafeec415..f586cbd30b77 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -11,7 +11,7 @@
  */
 
 #ifndef __DA732X_H_
-#define __DA732X_H
+#define __DA732X_H_
 
 #include <sound/soc.h>
 
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 6bc6efdec550..f1ec6e6bd08a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
 static const struct regmap_config rt5640_regmap = {
         .reg_bits = 8,
         .val_bits = 16,
+        .use_single_rw = true,
 
         .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) *
                                                RT5640_PR_SPACING),
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 67f14556462f..5337c448b5e3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
         { "BST2", NULL, "IN2P" },
         { "BST2", NULL, "IN2N" },
 
-        { "IN1P", NULL, "micbias1" },
-        { "IN1N", NULL, "micbias1" },
-        { "IN2P", NULL, "micbias1" },
-        { "IN2N", NULL, "micbias1" },
+        { "IN1P", NULL, "MICBIAS1" },
+        { "IN1N", NULL, "MICBIAS1" },
+        { "IN2P", NULL, "MICBIAS1" },
+        { "IN2N", NULL, "MICBIAS1" },
 
         { "ADC 1", NULL, "BST1" },
         { "ADC 1", NULL, "ADC 1 power" },
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 159e517fa09a..cef7776b712c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
         snd_soc_card_set_drvdata(&priv->snd_card, priv);
 
         ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
+        if (ret >= 0)
+                return ret;
 
 err:
         asoc_simple_card_unref(pdev);
         return ret;
 }
 
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+        return asoc_simple_card_unref(pdev);
+}
+
 static const struct of_device_id asoc_simple_of_match[] = {
         { .compatible = "simple-audio-card", },
         {},
@@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = {
                 .of_match_table = asoc_simple_of_match,
         },
         .probe = asoc_simple_card_probe,
+        .remove = asoc_simple_card_remove,
 };
 
 module_platform_driver(asoc_simple_card);
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index f8a6adc2d81c..4336d1831485 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
                 .stream_name = "TWL4030 Voice",
                 .cpu_dai_name = "omap-mcbsp.3",
                 .codec_dai_name = "twl4030-voice",
-                .platform_name = "omap-mcbsp.2",
+                .platform_name = "omap-mcbsp.3",
                 .codec_name = "twl4030-codec",
                 .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
                            SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 3fdf3be7b99a..f95e7ab135e8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv,
         };
 
         /* it shouldn't happen */
-        if (use_dvc & !use_src)
+        if (use_dvc && !use_src)
                 dev_err(dev, "DVC is selected without SRC\n");
 
         /* use SSIU or SSI ? */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4bfd4a9076f..889f4e3d35dc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
         device_initialize(rtd->dev);
         rtd->dev->parent = rtd->card->dev;
         rtd->dev->release = rtd_release;
-        rtd->dev->init_name = name;
+        dev_set_name(rtd->dev, "%s", name);
         dev_set_drvdata(rtd->dev, rtd);
         mutex_init(&rtd->pcm_mutex);
         INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 9577121ce971..ca8037634100 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -21,7 +21,7 @@
  */
 
 #ifndef __TEGRA_ASOC_UTILS_H__
-#define __TEGRA_ASOC_UTILS_H_
+#define __TEGRA_ASOC_UTILS_H__
 
 struct clk;
 struct device;