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

Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt

 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

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, \

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)

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];
 

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));
 

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),

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 */

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");

sound/soc/codecs/da732x.h

@@ -11,7 +11,7 @@
  */
 
 #ifndef __DA732X_H_
-#define __DA732X_H
+#define __DA732X_H_
 
 #include <sound/soc.h>
 

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),

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" },

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);

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,

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 ? */

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);

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;