Merge branch 'fix/hda' into topic/hda

Documentation/devicetree/bindings/sound/sgtl5000.txt

@@ -3,6 +3,8 @@
 Required properties:
 - compatible : "fsl,sgtl5000".
 
+- reg : the I2C address of the device
+
 Example:
 
 codec: sgtl5000@0a {

sound/pci/echoaudio/echoaudio_dsp.c

@@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip)
 	const struct firmware *fw;
 	int box_type, err;
 
-	if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
+	if (snd_BUG_ON(!chip->comm_page))
 		return -EPERM;
 
 	/* See if the ASIC is present and working - only if the DSP is already loaded */

sound/pci/hda/hda_codec.c

@@ -5497,10 +5497,6 @@ int snd_hda_suspend(struct hda_bus *bus)
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		if (hda_codec_is_power_on(codec))
 			hda_call_codec_suspend(codec);
-		else /* forcibly change the power to D3 even if not used */
-			hda_set_power_state(codec,
-					    codec->afg ? codec->afg : codec->mfg,
-					    AC_PWRST_D3);
 		if (codec->patch_ops.post_suspend)
 			codec->patch_ops.post_suspend(codec);
 	}

sound/pci/hda/hda_intel.c

@@ -2351,6 +2351,17 @@ static void azx_power_notify(struct hda_bus *bus)
  * power management
  */
 
+static int snd_hda_codecs_inuse(struct hda_bus *bus)
+{
+	struct hda_codec *codec;
+
+	list_for_each_entry(codec, &bus->codec_list, list) {
+		if (snd_hda_codec_needs_resume(codec))
+			return 1;
+	}
+	return 0;
+}
+
 static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
@@ -2397,7 +2408,8 @@ static int azx_resume(struct pci_dev *pci)
 		return -EIO;
 	azx_init_pci(chip);
 
-	azx_init_chip(chip, 1);
+	if (snd_hda_codecs_inuse(chip->bus))
+		azx_init_chip(chip, 1);
 
 	snd_hda_resume(chip->bus);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);

sound/pci/hda/patch_realtek.c

@@ -5381,6 +5381,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
 		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+	SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G",
+		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
 	SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
@@ -5414,6 +5416,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF),
 
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
+	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
@@ -5614,13 +5617,13 @@ static int patch_alc262(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
 	}
 #endif
-	alc_auto_parse_customize_define(codec);
-
 	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
 
 	alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc262_parse_auto_config(codec);
 	if (err < 0)
@@ -6083,6 +6086,7 @@ static const struct alc_fixup alc269_fixups[] = {
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
+	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
@@ -6222,8 +6226,6 @@ static int patch_alc269(struct hda_codec *codec)
 
 	spec->mixer_nid = 0x0b;
 
-	alc_auto_parse_customize_define(codec);
-
 	err = alc_codec_rename_from_preset(codec);
 	if (err < 0)
 		goto error;
@@ -6256,6 +6258,8 @@ static int patch_alc269(struct hda_codec *codec)
 		       alc269_fixup_tbl, alc269_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc269_parse_auto_config(codec);
 	if (err < 0)
@@ -6831,8 +6835,6 @@ static int patch_alc662(struct hda_codec *codec)
 	/* handle multiple HPs as is */
 	spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 
-	alc_auto_parse_customize_define(codec);
-
 	alc_fix_pll_init(codec, 0x20, 0x04, 15);
 
 	err = alc_codec_rename_from_preset(codec);
@@ -6849,6 +6851,9 @@ static int patch_alc662(struct hda_codec *codec)
 	alc_pick_fixup(codec, alc662_fixup_models,
 		       alc662_fixup_tbl, alc662_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc662_parse_auto_config(codec);
 	if (err < 0)

sound/pci/rme9652/hdsp.c

@@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 	strcpy(hw->name, "HDSP hwdep interface");
 
 	hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
+	hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl;
 
 	return 0;
 }

sound/soc/blackfin/bf5xx-ssm2602.c

@@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
 		.platform_name = "bfin-i2s-pcm-audio",
 		.codec_name = "ssm2602.0-001b",
 		.ops = &bf5xx_ssm2602_ops,
+		.dai_fmt = BF5XX_SSM2602_DAIFMT,
 	},
 	{
 		.name = "ssm2602",
@@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
 		.platform_name = "bfin-i2s-pcm-audio",
 		.codec_name = "ssm2602.0-001b",
 		.ops = &bf5xx_ssm2602_ops,
+		.dai_fmt = BF5XX_SSM2602_DAIFMT,
 	},
 };
 

sound/soc/codecs/cs42l73.c

@@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq)
 
 	/* MCLKX -> MCLK */
 	mclkx_coeff = cs42l73_get_mclkx_coeff(freq);
+	if (mclkx_coeff < 0)
+		return mclkx_coeff;
 
 	mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx /
 		cs42l73_mclkx_coeffs[mclkx_coeff].ratio;

sound/soc/codecs/tlv320aic23.c

@@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 				      enum snd_soc_bias_level level)
 {
-	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f;
+	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
@@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_OFF:
 		/* everything off, dac mute, inactive */
 		snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
-		snd_soc_write(codec, TLV320AIC23_PWR, 0xffff);
+		snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
 		break;
 	}
 	codec->dapm.bias_level = level;

sound/soc/codecs/wm8350.c

@@ -60,7 +60,7 @@ struct wm8350_jack_data {
 };
 
 struct wm8350_data {
-	struct snd_soc_codec codec;
+	struct wm8350 *wm8350;
 	struct wm8350_output out1;
 	struct wm8350_output out2;
 	struct wm8350_jack_data hpl;
@@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv,
 			   struct wm8350_jack_data *jack,
 			   u16 mask)
 {
-	struct wm8350 *wm8350 = priv->codec.control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	u16 reg;
 	int report;
 
@@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work)
 static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
 {
 	struct wm8350_data *priv = data;
-	struct wm8350 *wm8350 = priv->codec.control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	struct wm8350_jack_data *jack = NULL;
 
 	switch (irq - wm8350->irq_base) {
@@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
 static irqreturn_t wm8350_mic_handler(int irq, void *data)
 {
 	struct wm8350_data *priv = data;
-	struct wm8350 *wm8350 = priv->codec.control_data;
+	struct wm8350 *wm8350 = priv->wm8350;
 	u16 reg;
 	int report = 0;
 
@@ -1536,6 +1536,8 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 		return -ENOMEM;
 	snd_soc_codec_set_drvdata(codec, priv);
 
+	priv->wm8350 = wm8350;
+
 	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
 		priv->supplies[i].supply = supply_names[i];
 
@@ -1544,7 +1546,6 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
-	wm8350->codec.codec = codec;
 	codec->control_data = wm8350;
 
 	/* Put the codec into reset if it wasn't already */

sound/soc/codecs/wm_hubs.c

@@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 			    enum snd_soc_bias_level level)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
-	int val;
+	int mask, val;
 
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
@@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_ON:
 		/* Turn off any unneded single ended outputs */
 		val = 0;
+		mask = 0;
+
+		if (hubs->lineout1_se)
+			mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA;
+
+		if (hubs->lineout2_se)
+			mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA;
 
 		if (hubs->lineout1_se && hubs->lineout1n_ena)
 			val |= WM8993_LINEOUT1N_ENA;
@@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec,
 			val |= WM8993_LINEOUT2P_ENA;
 
 		snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3,
-				    WM8993_LINEOUT1N_ENA |
-				    WM8993_LINEOUT1P_ENA |
-				    WM8993_LINEOUT2N_ENA |
-				    WM8993_LINEOUT2P_ENA,
-				    val);
+				    mask, val);
 
 		/* Remove the input clamps */
 		snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG,

sound/soc/omap/omap-pcm.c

@@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	}
 
 out:
+	/* free preallocated buffers in case of error */
+	if (ret)
+		omap_pcm_free_dma_buffers(pcm);
+
 	return ret;
 }
 

sound/soc/samsung/s3c2412-i2s.c

@@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
 
 static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev)
 {
-	return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai);
+	return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai);
 }
 
 static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev)

sound/soc/sh/fsi.c

@@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data)
 	sg_dma_address(&sg) = buf;
 	sg_dma_len(&sg) = len;
 
-	desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir,
-						  DMA_PREP_INTERRUPT |
-						  DMA_CTRL_ACK);
+	desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir,
+				       DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
-		dev_err(dai->dev, "device_prep_slave_sg() fail\n");
+		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
 		return;
 	}
 

sound/soc/soc-core.c

@@ -3119,6 +3119,7 @@ int snd_soc_register_card(struct snd_soc_card *card)
 				 GFP_KERNEL);
 	if (card->rtd == NULL)
 		return -ENOMEM;
+	card->num_rtd = 0;
 	card->rtd_aux = &card->rtd[card->num_links];
 
 	for (i = 0; i < card->num_links; i++)
@@ -3630,10 +3631,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 	int i, ret;
 
 	num_routes = of_property_count_strings(np, propname);
-	if (num_routes & 1) {
+	if (num_routes < 0 || num_routes & 1) {
 		dev_err(card->dev,
-			"Property '%s's length is not even\n",
-			propname);
+		     "Property '%s' does not exist or its length is not even\n",
+		     propname);
 		return -EINVAL;
 	}
 	num_routes /= 2;

sound/soc/soc-dapm.c

@@ -67,6 +67,7 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_out_drv] = 10,
 	[snd_soc_dapm_hp] = 10,
 	[snd_soc_dapm_spk] = 10,
+	[snd_soc_dapm_line] = 10,
 	[snd_soc_dapm_post] = 11,
 };
 
@@ -75,6 +76,7 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_adc] = 1,
 	[snd_soc_dapm_hp] = 2,
 	[snd_soc_dapm_spk] = 2,
+	[snd_soc_dapm_line] = 2,
 	[snd_soc_dapm_out_drv] = 2,
 	[snd_soc_dapm_pga] = 4,
 	[snd_soc_dapm_mixer_named_ctl] = 5,