Merge tag 'v3.17-rc4' into asoc-simple

Linux 3.17-rc4

30 files changed, 160 insertions, 102 deletions

diff --git a/sound/core/info.c b/sound/core/info.c
index 051d55b..9f404e9 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 4560ca0..2c6fd80 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 },
 	},
 	/* FIXME: the following three formats are not defined properly yet */
 	[SNDRV_PCM_FORMAT_MPEG] = {
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index f96bf4c..95fc2eaf 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 d8ee7b0..4823c08 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 a9a30c0..e3a04d6 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 f2e34e3..5851249 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 07e7609..8371274 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 6f2fa838..6e5d0cb 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_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 36badba..99d7d7f 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info");
 #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec))
 
 #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882)
+#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883)
+#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec))
 
 struct hdmi_spec_per_cvt {
 	hda_nid_t cvt_nid;
@@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 			    mux_idx);
 
 	/* configure unused pins to choose other converters */
-	if (is_haswell_plus(codec) || is_valleyview(codec))
+	if (is_haswell_plus(codec) || is_valleyview_plus(codec))
 		intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx);
 
 	snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid);
@@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 		 *   and this can make HW reset converter selection on a pin.
 		 */
 		if (eld->eld_valid && !old_eld_valid && per_pin->setup) {
-			if (is_haswell_plus(codec) || is_valleyview(codec)) {
+			if (is_haswell_plus(codec) ||
+				is_valleyview_plus(codec)) {
 				intel_verify_pin_cvt_connect(codec, per_pin);
 				intel_not_share_assigned_cvt(codec, pin_nid,
 							per_pin->mux_idx);
@@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	bool non_pcm;
 	int pinctl;
 
-	if (is_haswell_plus(codec) || is_valleyview(codec)) {
+	if (is_haswell_plus(codec) || is_valleyview_plus(codec)) {
 		/* Verify pin:cvt selections to avoid silent audio after S3.
 		 * After S3, the audio driver restores pin:cvt selections
 		 * but this can happen before gfx is ready and such selection
@@ -2330,9 +2333,8 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		intel_haswell_fixup_enable_dp12(codec);
 	}
 
-	if (is_haswell(codec) || is_valleyview(codec)) {
+	if (is_haswell_plus(codec) || is_valleyview_plus(codec))
 		codec->depop_delay = 0;
-	}
 
 	if (hdmi_parse_codec(codec) < 0) {
 		codec->spec = NULL;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6b38ec3..1ba22fb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec)
 			    spec->pll_coef_idx);
 	val = snd_hda_codec_read(codec, spec->pll_nid, 0,
 				 AC_VERB_GET_PROC_COEF, 0);
+	if (val == -1)
+		return;
 	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
 			    spec->pll_coef_idx);
 	snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
@@ -326,6 +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:
@@ -2348,6 +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 */
@@ -2806,6 +2810,8 @@ static void alc286_shutup(struct hda_codec *codec)
 static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
 {
 	int val = alc_read_coef_idx(codec, 0x04);
+	if (val == -1)
+		return;
 	if (power_up)
 		val |= 1 << 11;
 	else
@@ -3264,6 +3270,15 @@ static int alc269_resume(struct hda_codec *codec)
 	snd_hda_codec_resume_cache(codec);
 	alc_inv_dmic_sync(codec, true);
 	hda_call_check_power_status(codec, 0x01);
+
+	/* on some machine, the BIOS will clear the codec gpio data when enter
+	 * suspend, and won't restore the data after resume, so we restore it
+	 * in the driver.
+	 */
+	if (spec->gpio_led)
+		snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA,
+			    spec->gpio_led);
+
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
 
@@ -4395,6 +4410,7 @@ 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[] = {
@@ -4842,6 +4858,22 @@ 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 },
+			{ },
+		},
+	},
 
 };
 
@@ -4853,6 +4885,7 @@ 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, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
@@ -5311,27 +5344,30 @@ static void alc269_fill_coef(struct hda_codec *codec)
 	if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
 		val = alc_read_coef_idx(codec, 0x04);
 		/* Power up output pin */
-		alc_write_coef_idx(codec, 0x04, val | (1<<11));
+		if (val != -1)
+			alc_write_coef_idx(codec, 0x04, val | (1<<11));
 	}
 
 	if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		val = alc_read_coef_idx(codec, 0xd);
-		if ((val & 0x0c00) >> 10 != 0x1) {
+		if (val != -1 && (val & 0x0c00) >> 10 != 0x1) {
 			/* Capless ramp up clock control */
 			alc_write_coef_idx(codec, 0xd, val | (1<<10));
 		}
 		val = alc_read_coef_idx(codec, 0x17);
-		if ((val & 0x01c0) >> 6 != 0x4) {
+		if (val != -1 && (val & 0x01c0) >> 6 != 0x4) {
 			/* Class D power on reset */
 			alc_write_coef_idx(codec, 0x17, val | (1<<7));
 		}
 	}
 
 	val = alc_read_coef_idx(codec, 0xd); /* Class D */
-	alc_write_coef_idx(codec, 0xd, val | (1<<14));
+	if (val != -1)
+		alc_write_coef_idx(codec, 0xd, val | (1<<14));
 
 	val = alc_read_coef_idx(codec, 0x4); /* HP */
-	alc_write_coef_idx(codec, 0x4, val | (1<<11));
+	if (val != -1)
+		alc_write_coef_idx(codec, 0x4, val | (1<<11));
 }
 
 /*
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index bd41ee4..2c71f16 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	else
 		rates = &arizona_48k_bclk_rates[0];
 
+	wl = snd_pcm_format_width(params_format(params));
+
 	if (tdm_slots) {
 		arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
 				tdm_slots, tdm_width);
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		channels = tdm_slots;
 	} else {
 		bclk_target = snd_soc_params_to_bclk(params);
+		tdm_width = wl;
 	}
 
 	if (chan_limit && chan_limit < channels) {
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
 			rates[bclk], rates[bclk] / lrclk);
 
-	wl = snd_pcm_format_width(params_format(params));
-	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width;
 
 	reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
 
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30c..9852320 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 1dceafe..f586cbd 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/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 163ec38..0c8aefa 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds =
 			pcm512x_ramp_step_text);
 
 static const struct snd_kcontrol_new pcm512x_controls[] = {
-SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2,
 		 PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
 SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
 	       PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
 SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
 	       PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
-SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
 	   PCM512x_RQMR_SHIFT, 1, 1),
 
 SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 6bc6efd..f1ec6e6 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 67f1455..5337c44 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/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c28508d..6a6b2ff 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -403,7 +403,8 @@ out:
 	return ret;
 }
 
-static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+				      int div, bool explicit)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
@@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
 		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
 			       ACLKRDIV(div - 1), ACLKRDIV_MASK);
-		mcasp->bclk_div = div;
+		if (explicit)
+			mcasp->bclk_div = div;
 		break;
 
 	case 2:		/* BCLK/LRCLK ratio */
@@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
 	return 0;
 }
 
+static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+				    int div)
+{
+	return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+}
+
 static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 				    unsigned int freq, int dir)
 {
@@ -738,7 +746,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 				 "Inaccurate BCLK: %u Hz / %u != %u Hz\n",
 				 mcasp->sysclk_freq, div, bclk_freq);
 		}
-		davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
+		__davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
 	}
 
 	ret = mcasp_common_hw_param(mcasp, substream->stream,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f54a8fc..f3012b6 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI
 	tristate "Enhanced Serial Audio Interface (ESAI) module support"
 	select REGMAP_MMIO
 	select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
-	select SND_SOC_FSL_UTILS
 	help
 	  Say Y if you want to add Enhanced Synchronous Audio Interface
 	  (ESAI) support for the Freescale CPUs.
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e..a3b29ed 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -18,7 +18,6 @@
 
 #include "fsl_esai.h"
 #include "imx-pcm.h"
-#include "fsl_utils.h"
 
 #define FSL_ESAI_RATES		SNDRV_PCM_RATE_8000_192000
 #define FSL_ESAI_FORMATS	(SNDRV_PCM_FMTBIT_S8 | \
@@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = {
 	.hw_params = fsl_esai_hw_params,
 	.set_sysclk = fsl_esai_set_dai_sysclk,
 	.set_fmt = fsl_esai_set_dai_fmt,
-	.xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
 	.set_tdm_slot = fsl_esai_set_dai_tdm_slot,
 };
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index a887707..709ce67 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -492,12 +492,19 @@ 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", },
 	{},
@@ -511,6 +518,7 @@ 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/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
index 42edc6f..03d0a16 100644
--- a/sound/soc/intel/sst-acpi.c
+++ b/sound/soc/intel/sst-acpi.c
@@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
 };
 
 static struct sst_acpi_mach baytrail_machines[] = {
-	{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
-	{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" },
+	{ "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
+	{ "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
 	{}
 };
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index 67673a2..b4ad98c 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = {
 	.ops = &sst_byt_ops,
 };
 
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
+int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
 {
 	struct sst_byt *byt = pdata->dsp;
 
@@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
 	sst_byt_drop_all(byt);
 	dev_dbg(byt->dev, "dsp in reset\n");
 
-	return 0;
-}
-EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq);
-
-int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
-{
-	struct sst_byt *byt = pdata->dsp;
-
 	dev_dbg(byt->dev, "free all blocks and unload fw\n");
 	sst_fw_unload(byt->fw);
 
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
index 06a4d20..8faff6d 100644
--- a/sound/soc/intel/sst-baytrail-ipc.h
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt,
 int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
 void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
 struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata);
 int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 599401c..eab1c7d 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -59,6 +59,9 @@ struct sst_byt_priv_data {
 
 	/* DAI data */
 	struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+
+	/* flag indicating is stream context restore needed after suspend */
+	bool restore_stream;
 };
 
 /* this may get called several times by oss emulation */
@@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		sst_byt_stream_start(byt, pcm_data->stream, 0);
 		break;
 	case SNDRV_PCM_TRIGGER_RESUME:
-		schedule_work(&pcm_data->work);
+		if (pdata->restore_stream == true)
+			schedule_work(&pcm_data->work);
+		else
+			sst_byt_stream_resume(byt, pcm_data->stream);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		sst_byt_stream_resume(byt, pcm_data->stream);
@@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		sst_byt_stream_stop(byt, pcm_data->stream);
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
+		pdata->restore_stream = false;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		sst_byt_stream_pause(byt, pcm_data->stream);
 		break;
@@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = {
 };
 
 #ifdef CONFIG_PM
-static int sst_byt_pcm_dev_suspend_noirq(struct device *dev)
-{
-	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-	int ret;
-
-	dev_dbg(dev, "suspending noirq\n");
-
-	/* at this point all streams will be stopped and context saved */
-	ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata);
-	if (ret < 0) {
-		dev_err(dev, "failed to suspend %d\n", ret);
-		return ret;
-	}
-
-	return ret;
-}
-
 static int sst_byt_pcm_dev_suspend_late(struct device *dev)
 {
 	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+	struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev);
 	int ret;
 
 	dev_dbg(dev, "suspending late\n");
@@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev)
 		return ret;
 	}
 
+	priv_data->restore_stream = true;
+
 	return ret;
 }
 
 static int sst_byt_pcm_dev_resume_early(struct device *dev)
 {
 	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+	int ret;
 
 	dev_dbg(dev, "resume early\n");
 
 	/* load fw and boot DSP */
-	return sst_byt_dsp_boot(dev, sst_pdata);
-}
-
-static int sst_byt_pcm_dev_resume(struct device *dev)
-{
-	struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-
-	dev_dbg(dev, "resume\n");
+	ret = sst_byt_dsp_boot(dev, sst_pdata);
+	if (ret)
+		return ret;
 
 	/* wait for FW to finish booting */
 	return sst_byt_dsp_wait_for_ready(dev, sst_pdata);
 }
 
 static const struct dev_pm_ops sst_byt_pm_ops = {
-	.suspend_noirq = sst_byt_pcm_dev_suspend_noirq,
 	.suspend_late = sst_byt_pcm_dev_suspend_late,
 	.resume_early = sst_byt_pcm_dev_resume_early,
-	.resume = sst_byt_pcm_dev_resume,
 };
 
 #define SST_BYT_PM_OPS	(&sst_byt_pm_ops)
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index f8a6adc..4336d18 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/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 0109f6c2..a8e0974 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
 			  SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 |	\
 			  SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
-			    SNDRV_PCM_FMTBIT_S24_LE |	\
-			    SNDRV_PCM_FMTBIT_S32_LE)
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
 	.startup	= pxa_ssp_startup,
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 3fdf3be..f95e7ab 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 d4bfd4a..889f4e3 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/soc-dapm.c b/sound/soc/soc-dapm.c
index 8348352..177bd86 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 	struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	unsigned int reg_val, val;
-	int ret = 0;
 
-	if (e->reg != SND_SOC_NOPM)
-		ret = soc_dapm_read(dapm, e->reg, &reg_val);
-	else
+	if (e->reg != SND_SOC_NOPM) {
+		int ret = soc_dapm_read(dapm, e->reg, &reg_val);
+		if (ret)
+			return ret;
+	} else {
 		reg_val = dapm_kcontrol_get_value(kcontrol);
+	}
 
 	val = (reg_val >> e->shift_l) & e->mask;
 	ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
@@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
 		ucontrol->value.enumerated.item[1] = val;
 	}
 
-	return ret;
+	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
 
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 9577121..ca80376 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;