Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This contains slightly more volumes than usual at this stage, mostly
  because of my vacation in the last week.  Nothing to scare, all small
  and/or trivial fixes:

   - Fix loop path handling in ASoC DAPM
   - Some memory handling fixes in ASoC core
   - Fix spear_pcm to adapt to the updated API
   - HD-audio HDMI ELD handling fixes
   - Fix for CM6331 USB-audio SRC change bugs
   - Revert power_save_controller option change due to user-space usage
   - A few other small ASoC and HD-audio fixes"

* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/generic - fix uninitialized variable
  Revert "ALSA: hda - Allow power_save_controller option override DCAPS"
  ALSA: hda - fix typo in proc output
  ALSA: hda - Enabling Realtek ALC 671 codec
  ALSA: usb: Work around CM6631 sample rate change bug
  ALSA: hda - bug fix on HDMI ELD debug message
  ALSA: hda - bug fix on return value when getting HDMI ELD info
  ASoC: dma-sh7760: Fix compile error
  ASoC: core: fix invalid free of devm_ allocated data
  ASoC: spear_pcm: Update to new pcm_new() API
  ASoC:: max98090: Remove executable bit
  ASoC: dapm: Fix pointer dereference in is_connected_output_ep()
  ASoC: pcm030 audio fabric: remove __init from probe
  ASoC: imx-ssi: Fix occasional AC97 reset failure
  ASoC: core: fix possible memory leak in snd_soc_bytes_put()
  ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff()
  ASoC: dapm: Fix handling of loops
  ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case

20 files changed, 83 insertions, 37 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 4499bd9..95731a0 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -890,9 +890,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     enable_msi	- Enable Message Signaled Interrupt (MSI) (default = off)
     power_save	- Automatic power-saving timeout (in second, 0 =
 		disable)
-    power_save_controller - Support runtime D3 of HD-audio controller
-		(-1 = on for supported chip (default), false = off,
-		 true = force to on even for unsupported hardware)
+    power_save_controller - Reset HD-audio controller in power-saving mode
+		(default = on)
     align_buffer_size - Force rounding of buffer/period sizes to multiples
     		      of 128 bytes. This is more efficient in terms of memory
 		      access but isn't required by the HDA spec and prevents
diff --git a/include/sound/max98090.h b/include/sound/max98090.h
index 95efb13..95efb13 100755..100644
--- a/include/sound/max98090.h
+++ b/include/sound/max98090.h
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e1ef63d..44a30b1 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -488,6 +488,7 @@ struct snd_soc_dapm_path {
 	/* status */
 	u32 connect:1;	/* source and sink widgets are connected */
 	u32 walked:1;	/* path has been walked */
+	u32 walking:1;  /* path is in the process of being walked */
 	u32 weak:1;	/* path ignored for power management */
 
 	int (*connected)(struct snd_soc_dapm_widget *source,
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index ecdf30e..4aba764 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg)
 		"Line Out", "Speaker", "HP Out", "CD",
 		"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
 		"Line In", "Aux", "Mic", "Telephony",
-		"SPDIF In", "Digitial In", "Reserved", "Other"
+		"SPDIF In", "Digital In", "Reserved", "Other"
 	};
 
 	return jack_types[(cfg & AC_DEFCFG_DEVICE)
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 7dd8463..d0d7ac1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
 		     unsigned char *buf, int *eld_size)
 {
 	int i;
-	int ret;
+	int ret = 0;
 	int size;
 
 	/*
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 43c2ea5..2dbe767 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path);
 static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
 {
 	struct hda_gen_spec *spec = codec->spec;
-	bool changed;
+	bool changed = false;
 	int i;
 
 	if (!spec->power_down_unused || path->active)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 418bfc0..bcd40ee 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
  * this may give more power-saving, but will take longer time to
  * wake up.
  */
-static int power_save_controller = -1;
-module_param(power_save_controller, bint, 0644);
+static bool power_save_controller = 1;
+module_param(power_save_controller, bool, 0644);
 MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
 #endif /* CONFIG_PM */
 
@@ -2931,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
-	if (power_save_controller > 0)
-		return 0;
 	if (!power_save_controller ||
 	    !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
 		return -EBUSY;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 78e1827..de8ac5c 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 
 	_snd_printd(SND_PR_VERBOSE,
 		"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
+		codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid);
 
 	if (eld->eld_valid) {
 		if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 563c24d..f15c36b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	const hda_nid_t *ssids;
 
 	if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
-	    codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
+	    codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 ||
+	    codec->vendor_id == 0x10ec0671)
 		ssids = alc663_ssids;
 	else
 		ssids = alc662_ssids;
@@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
 	{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
 	{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
+	{ .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
 	{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
 	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index fc17604..fc17604 100755..100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 7e103f2..7e103f2 100755..100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index f2d61a1..566ea32 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		width = SI476X_PCM_FORMAT_S8;
+		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		width = SI476X_PCM_FORMAT_S16_LE;
 		break;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f3f7e75..9af1bdd 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 						&buf_list);
 			if (!buf) {
 				adsp_err(dsp, "Out of memory\n");
-				return -ENOMEM;
+				ret = -ENOMEM;
+				goto out_fw;
 			}
 
 			adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
@@ -865,7 +866,7 @@ out_fw:
 	wm_adsp_buf_free(&buf_list);
 out:
 	kfree(file);
-	return 0;
+	return ret;
 }
 
 int wm_adsp1_init(struct wm_adsp *adsp)
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 55464a5..810c7ee 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97)
 
 	if (imx_ssi->ac97_reset)
 		imx_ssi->ac97_reset(ac97);
+	/* First read sometimes fails, do a dummy read */
+	imx_ssi_ac97_read(ac97, 0);
 }
 
 static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
@@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
 
 	if (imx_ssi->ac97_warm_reset)
 		imx_ssi->ac97_warm_reset(ac97);
+
+	/* First read sometimes fails, do a dummy read */
+	imx_ssi_ac97_read(ac97, 0);
 }
 
 struct snd_ac97_bus_ops soc_ac97_ops = {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 8e52c14..eb43738 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = {
 	.num_links = ARRAY_SIZE(pcm030_fabric_dai),
 };
 
-static int __init pcm030_fabric_probe(struct platform_device *op)
+static int pcm030_fabric_probe(struct platform_device *op)
 {
 	struct device_node *np = op->dev.of_node;
 	struct device_node *platform_np;
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 19eff8f..1a8b03e 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static struct snd_soc_platform sh7760_soc_platform = {
-	.pcm_ops 	= &camelot_pcm_ops,
+static struct snd_soc_platform_driver sh7760_soc_platform = {
+	.ops		= &camelot_pcm_ops,
 	.pcm_new	= camelot_pcm_new,
 	.pcm_free	= camelot_pcm_free,
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b7e84a7..507d251 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 	if (params->mask) {
 		ret = regmap_read(codec->control_data, params->base, &val);
 		if (ret != 0)
-			return ret;
+			goto out;
 
 		val &= params->mask;
 
@@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 			((u32 *)data)[0] |= cpu_to_be32(val);
 			break;
 		default:
-			return -EINVAL;
+			ret = -EINVAL;
+			goto out;
 		}
 	}
 
 	ret = regmap_raw_write(codec->control_data, params->base,
 			       data, len);
 
+out:
 	kfree(data);
 
 	return ret;
@@ -4197,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			dev_err(card->dev,
 				"ASoC: Property '%s' index %d could not be read: %d\n",
 				propname, 2 * i, ret);
-			kfree(routes);
 			return -EINVAL;
 		}
 		ret = of_property_read_string_index(np, propname,
@@ -4206,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			dev_err(card->dev,
 				"ASoC: Property '%s' index %d could not be read: %d\n",
 				propname, (2 * i) + 1, ret);
-			kfree(routes);
 			return -EINVAL;
 		}
 	}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d6a9b3..d6d9ba2 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 		if (path->weak)
 			continue;
 
+		if (path->walking)
+			return 1;
+
 		if (path->walked)
 			continue;
 
@@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 
 		if (path->sink && path->connect) {
 			path->walked = 1;
+			path->walking = 1;
 
 			/* do we need to add this widget to the list ? */
 			if (list) {
@@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 					dev_err(widget->dapm->dev,
 						"ASoC: could not add widget %s\n",
 						widget->name);
+					path->walking = 0;
 					return con;
 				}
 			}
 
 			con += is_connected_output_ep(path->sink, list);
+
+			path->walking = 0;
 		}
 	}
 
@@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 		if (path->weak)
 			continue;
 
+		if (path->walking)
+			return 1;
+
 		if (path->walked)
 			continue;
 
@@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 
 		if (path->source && path->connect) {
 			path->walked = 1;
+			path->walking = 1;
 
 			/* do we need to add this widget to the list ? */
 			if (list) {
@@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 					dev_err(widget->dapm->dev,
 						"ASoC: could not add widget %s\n",
 						widget->name);
+					path->walking = 0;
 					return con;
 				}
 			}
 
 			con += is_connected_input_ep(path->source, list);
+
+			path->walking = 0;
 		}
 	}
 
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 9b76cc5..5e7aebe 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm)
 
 static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
 
-static int spear_pcm_new(struct snd_card *card,
-		struct snd_soc_dai *dai, struct snd_pcm *pcm)
+static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_card *card = rtd->card->snd_card;
 	int ret;
 
 	if (!card->dev->dma_mask)
@@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
-		ret = spear_pcm_preallocate_dma_buffer(pcm,
+	if (rtd->cpu_dai->driver->playback.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
 				SNDRV_PCM_STREAM_PLAYBACK,
 				spear_pcm_hardware.buffer_bytes_max);
 		if (ret)
 			return ret;
 	}
 
-	if (dai->driver->capture.channels_min) {
-		ret = spear_pcm_preallocate_dma_buffer(pcm,
+	if (rtd->cpu_dai->driver->capture.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(rtd->pcm,
 				SNDRV_PCM_STREAM_CAPTURE,
 				spear_pcm_hardware.buffer_bytes_max);
 		if (ret)
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 5e634a2..9e2703a 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 {
 	struct usb_device *dev = chip->dev;
 	unsigned char data[4];
-	int err, crate;
+	int err, cur_rate, prev_rate;
 	int clock = snd_usb_clock_find_source(chip, fmt->clock);
 
 	if (clock < 0)
@@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 		return -ENXIO;
 	}
 
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
+			      data, sizeof(data));
+	if (err < 0) {
+		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+			   dev->devnum, iface, fmt->altsetting);
+		prev_rate = 0;
+	} else {
+		prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
+	}
+
 	data[0] = rate;
 	data[1] = rate >> 8;
 	data[2] = rate >> 16;
@@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
 		return err;
 	}
 
-	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
-				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8,
-				   snd_usb_ctrl_intf(chip) | (clock << 8),
-				   data, sizeof(data))) < 0) {
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8,
+			      snd_usb_ctrl_intf(chip) | (clock << 8),
+			      data, sizeof(data));
+	if (err < 0) {
 		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);
-		return err;
+		cur_rate = 0;
+	} else {
+		cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
 	}
 
-	crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
-	if (crate != rate)
-		snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+	if (cur_rate != rate) {
+		snd_printd(KERN_WARNING
+			   "current rate %d is different from the runtime rate %d\n",
+			   cur_rate, rate);
+	}
+
+	/* Some devices doesn't respond to sample rate changes while the
+	 * interface is active. */
+	if (rate != prev_rate) {
+		usb_set_interface(dev, iface, 0);
+		usb_set_interface(dev, iface, fmt->altsetting);
+	}
 
 	return 0;
 }