diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt
index 46f344965313f1081dbd7894e738795a8b935815..4eb7997674a09006dfa0a991b634082ddc64974d 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 be6ecae247b0c608df5aa2554079d6bc4efca482..c83a334dd00fa9186404917183115fca3124670c 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 f96bf4c7c23236a773f13a0c318ddb5b6f275977..95fc2eaf11dc0f91bcb235a2929b300c39448052 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 d8ee7b0e938629a828571fcd45ce5da5edc02469..4823c08196ac775f150656b712dd7839b5428a29 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 a9a30c0161f17be87e9c79403d1c913f85ab4c86..e3a04d69c85363dfffdc6bebf5400b95cff401fa 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 6f2fa838b63532852ab535ab9ff189b99c52edea..6e5d0cb4e3d737ff43f0eb63e298de817552d224 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 d446ac3137b37ac6f3512f8fa7830b537632bca8..1ba22fb527c288b751d3064e041759e881013605 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 a20b30ca52c09dc13507d55afef7854a23f29e40..98523209f7391852cb0e517fb373c6635e2ded99 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 1dceafeec4151b4ae85b629aaa61e0737de73c34..f586cbd30b776ead146eb7f0edb4d11f7c74f013 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 6bc6efdec550db684e4f8bfa32bd88107902d9cd..f1ec6e6bd08aced4be7c38d868ccba94a655c5f7 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 67f14556462ff35bd6601062fc55a40ec2a96310..5337c448b5e365de204a95e3a945291dd228844b 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 159e517fa09aa6ee5d061e658cafcd9d0f8f12a5..cef7776b712cff43f37f81648a6fe2a59256ddd9 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 f8a6adc2d81ca8cecec98a3b768cbc1ccef9b0c0..4336d1831485f77787f384d77ad4785fce6ad02a 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 3fdf3be7b99ab558e41fd26080b3bb80c5d53b59..f95e7ab135e83266f958c49801284a6e780744a3 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 d4bfd4a9076fe02fd50495e0fee6f97471500bb5..889f4e3d35dc53548f59f159666fbc668f7bd39b 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 9577121ce971247407b6d04957de548a5b3fd67d..ca8037634100b0a814927f52da9d77fca28fd657 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;