Merge branch 'for-linus' into for-next

Merging for-linus branch for syncing the latest STAC/IDT codec
changes to be affected by the upcoming hda-jack rewrites.

19 files changed, 101 insertions, 39 deletions

diff --git a/sound/core/info.c b/sound/core/info.c
index 051d55b05521..9f404e965ea2 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -684,7 +684,7 @@ int snd_info_card_free(struct snd_card *card)
  * snd_info_get_line - read one line from the procfs buffer
  * @buffer: the procfs buffer
  * @line: the buffer to store
- * @len: the max. buffer size - 1
+ * @len: the max. buffer size
  *
  * Reads one line from the buffer and stores the string.
  *
@@ -704,7 +704,7 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
 			buffer->stop = 1;
 		if (c == '\n')
 			break;
-		if (len) {
+		if (len > 1) {
 			len--;
 			*line++ = c;
 		}
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 83f54e1dbac9..ae7a0feb3b76 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -142,11 +142,11 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = {
 	},
 	[SNDRV_PCM_FORMAT_DSD_U8] = {
 		.width = 8, .phys = 8, .le = 1, .signd = 0,
-		.silence = {},
+		.silence = { 0x69 },
 	},
 	[SNDRV_PCM_FORMAT_DSD_U16_LE] = {
 		.width = 16, .phys = 16, .le = 1, .signd = 0,
-		.silence = {},
+		.silence = { 0x69, 0x69 },
 	},
 	[SNDRV_PCM_FORMAT_DSD_U32_LE] = {
 		.width = 32, .phys = 32, .le = 1, .signd = 0,
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/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h
index f2e34e3f27ee..5851249f11d9 100644
--- a/sound/pci/ctxfi/ct20k1reg.h
+++ b/sound/pci/ctxfi/ct20k1reg.h
@@ -7,7 +7,7 @@
  */
 
 #ifndef CT20K1REG_H
-#define CT20k1REG_H
+#define CT20K1REG_H
 
 /* 20k1 registers */
 #define 	DSPXRAM_START 			0x000000
@@ -632,5 +632,3 @@
 #define I2SD_R    0x19L
 
 #endif /* CT20K1REG_H */
-
-
diff --git a/sound/pci/hda/ca0132_regs.h b/sound/pci/hda/ca0132_regs.h
index 07e760937d3c..8371274aa811 100644
--- a/sound/pci/hda/ca0132_regs.h
+++ b/sound/pci/hda/ca0132_regs.h
@@ -20,7 +20,7 @@
  */
 
 #ifndef __CA0132_REGS_H
-#define __CA0312_REGS_H
+#define __CA0132_REGS_H
 
 #define DSP_CHIP_OFFSET                0x100000
 #define DSP_DBGCNTL_MODULE_OFFSET      0xE30
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c0b03c112187..e0c5bc1d671b 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -216,6 +216,7 @@ 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,
@@ -663,6 +664,12 @@ 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,
@@ -743,7 +750,7 @@ 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 cbc4d25e4538..6b1a5de07e35 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -373,6 +373,7 @@ 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:
@@ -2346,6 +2347,7 @@ static int patch_alc882(struct hda_codec *codec)
 	switch (codec->vendor_id) {
 	case 0x10ec0882:
 	case 0x10ec0885:
+	case 0x10ec0900:
 		break;
 	default:
 		/* ALC883 and variants */
@@ -4353,6 +4355,7 @@ enum {
 	ALC292_FIXUP_TPT440_DOCK,
 	ALC283_FIXUP_BXBT2807_MIC,
 	ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED,
+	ALC282_FIXUP_ASPIRE_V5_PINS,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -4800,6 +4803,22 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained_before = true,
 		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
+	[ALC282_FIXUP_ASPIRE_V5_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x12, 0x90a60130 },
+			{ 0x14, 0x90170110 },
+			{ 0x17, 0x40000008 },
+			{ 0x18, 0x411111f0 },
+			{ 0x19, 0x411111f0 },
+			{ 0x1a, 0x411111f0 },
+			{ 0x1b, 0x411111f0 },
+			{ 0x1d, 0x40f89b2d },
+			{ 0x1e, 0x411111f0 },
+			{ 0x21, 0x0321101f },
+			{ },
+		},
+	},
 
 };
 
@@ -4811,6 +4830,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
 	SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
+	SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER),
 	SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f26ec04a29b5..60aebd0f5e56 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -565,8 +565,8 @@ static void stac_init_power_map(struct hda_codec *codec)
 		if (snd_hda_jack_tbl_get(codec, nid))
 			continue;
 		if (def_conf == AC_JACK_PORT_COMPLEX &&
-		    !(spec->vref_mute_led_nid == nid ||
-		      is_jack_detectable(codec, nid))) {
+		    spec->vref_mute_led_nid != nid &&
+		    is_jack_detectable(codec, nid)) {
 			snd_hda_jack_detect_enable_callback(codec, nid,
 							    STAC_PWR_EVENT,
 							    jack_update_power);
@@ -4272,11 +4272,18 @@ static int stac_parse_auto_config(struct hda_codec *codec)
 			return err;
 	}
 
-	stac_init_power_map(codec);
-
 	return 0;
 }
 
+static int stac_build_controls(struct hda_codec *codec)
+{
+	int err = snd_hda_gen_build_controls(codec);
+
+	if (err < 0)
+		return err;
+	stac_init_power_map(codec);
+	return 0;
+}
 
 static int stac_init(struct hda_codec *codec)
 {
@@ -4388,7 +4395,7 @@ static int stac_suspend(struct hda_codec *codec)
 #endif /* CONFIG_PM */
 
 static const struct hda_codec_ops stac_patch_ops = {
-	.build_controls = snd_hda_gen_build_controls,
+	.build_controls = stac_build_controls,
 	.build_pcms = snd_hda_gen_build_pcms,
 	.init = stac_init,
 	.free = stac_free,
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;