Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix mute control with some ALC262 models
  ALSA: snd_usb_caiaq: add support for Audio2DJ
  ALSA: pcm - Fix hwptr buffer-size overlap bug
  ALSA: pcm - Fix warnings in debug loggings
  ALSA: pcm - Add logging of hwptr updates and interrupt updates
  ASoC: tlv320aic3x: Enable PLL when not bypassed
  ALSA: hda - Restore GPIO1 properly at resume with AD1984A
  ALSA: ctxfi - Fix uninitialized error checks
  ALSA: hda - Use snprintf() to be safer
  ALSA: usb-audio - Volume control quirk for QuickCam E 3500
  ALSA: pcm - Fix regressions with VMware

Documentation/sound/alsa/Procfile.txt

@@ -101,6 +101,8 @@ card*/pcm*/xrun_debug
 	  bit 0 = Enable XRUN/jiffies debug messages
 	  bit 1 = Show stack trace at XRUN / jiffies check
 	  bit 2 = Enable additional jiffies check
+	  bit 3 = Log hwptr update at each period interrupt
+	  bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr()
 
 	When the bit 0 is set, the driver will show the messages to
 	kernel log when an xrun is detected.  The debug message is
@@ -117,6 +119,9 @@ card*/pcm*/xrun_debug
 	buggy) hardware that doesn't give smooth pointer updates.
 	This feature is enabled via the bit 2.
 
+	Bits 3 and 4 are for logging the hwptr records.  Note that
+	these will give flood of kernel messages.
+
 card*/pcm*/sub*/info
 	The general information of this PCM sub-stream.
 

sound/core/pcm_lib.c

@@ -233,6 +233,18 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 		xrun(substream);
 		return -EPIPE;
 	}
+	if (xrun_debug(substream, 8)) {
+		char name[16];
+		pcm_debug_name(substream, name, sizeof(name));
+		snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, "
+			   "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+			   name, (unsigned int)pos,
+			   (unsigned int)runtime->period_size,
+			   (unsigned int)runtime->buffer_size,
+			   (unsigned long)old_hw_ptr,
+			   (unsigned long)runtime->hw_ptr_base,
+			   (unsigned long)runtime->hw_ptr_interrupt);
+	}
 	hw_base = runtime->hw_ptr_base;
 	new_hw_ptr = hw_base + pos;
 	hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
@@ -244,6 +256,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 			delta = new_hw_ptr - hw_ptr_interrupt;
 	}
 	if (delta < 0) {
+		if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr)
 			delta += runtime->buffer_size;
 		if (delta < 0) {
 			hw_ptr_error(substream, 
@@ -251,11 +264,19 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 				     "(stream=%i, pos=%ld, intr_ptr=%ld)\n",
 				     substream->stream, (long)pos,
 				     (long)hw_ptr_interrupt);
+#if 1
+			/* simply skipping the hwptr update seems more
+			 * robust in some cases, e.g. on VMware with
+			 * inaccurate timer source
+			 */
+			return 0; /* skip this update */
+#else
 			/* rebase to interrupt position */
 			hw_base = new_hw_ptr = hw_ptr_interrupt;
 			/* align hw_base to buffer_size */
 			hw_base -= hw_base % runtime->buffer_size;
 			delta = 0;
+#endif
 		} else {
 			hw_base += runtime->buffer_size;
 			if (hw_base >= runtime->boundary)
@@ -344,6 +365,19 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
 		xrun(substream);
 		return -EPIPE;
 	}
+	if (xrun_debug(substream, 16)) {
+		char name[16];
+		pcm_debug_name(substream, name, sizeof(name));
+		snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, "
+			   "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n",
+			   name, (unsigned int)pos,
+			   (unsigned int)runtime->period_size,
+			   (unsigned int)runtime->buffer_size,
+			   (unsigned long)old_hw_ptr,
+			   (unsigned long)runtime->hw_ptr_base,
+			   (unsigned long)runtime->hw_ptr_interrupt);
+	}
+
 	hw_base = runtime->hw_ptr_base;
 	new_hw_ptr = hw_base + pos;
 

sound/pci/ctxfi/ctamixer.c

@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,
 
 	/* Allocate mem for amixer resource */
 	amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
-	if (NULL == amixer) {
+	if (!amixer)
-		err = -ENOMEM;
+		return -ENOMEM;
-		return err;
-	}
 
 	/* Check whether there are sufficient
 	 * amixer resources to meet request. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,
 
 	/* Allocate mem for sum resource */
 	sum = kzalloc(sizeof(*sum), GFP_KERNEL);
-	if (NULL == sum) {
+	if (!sum)
-		err = -ENOMEM;
+		return -ENOMEM;
-		return err;
-	}
 
 	/* Check whether there are sufficient sum resources to meet request. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);

sound/pci/ctxfi/ctsrc.c

@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,
 
 	/* Allocate mem for SRCIMP resource */
 	srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
-	if (NULL == srcimp) {
+	if (!srcimp)
-		err = -ENOMEM;
+		return -ENOMEM;
-		return err;
-	}
 
 	/* Check whether there are sufficient SRCIMP resources. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);

sound/pci/hda/patch_analog.c

@@ -3754,7 +3754,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
 	int mute = (!ucontrol->value.integer.value[0] &&
 		    !ucontrol->value.integer.value[1]);
 	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
 			    mute ? 0x02 : 0x0);
 	return ret;
 }

sound/pci/hda/patch_realtek.c

@@ -10631,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
 	alc262_lenovo_3000_automute(codec, 1);
 }
 
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+				  int dir, int idx, long *valp)
+{
+	int i, change = 0;
+
+	for (i = 0; i < 2; i++, valp++)
+		change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+						   HDA_AMP_MUTE,
+						   *valp ? 0 : HDA_AMP_MUTE);
+	return change;
+}
+
 /* bind hp and internal speaker mute (with plug check) */
 static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
@@ -10639,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
-						 HDA_AMP_MUTE,
+	change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
-						 valp ? 0 : HDA_AMP_MUTE);
-	change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-
 	if (change)
 		alc262_fujitsu_automute(codec, 0);
 	return change;
@@ -10680,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+	change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-
 	if (change)
 		alc262_lenovo_3000_automute(codec, 0);
 	return change;
@@ -11854,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
+	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
-					  HDA_AMP_MUTE,
-					  valp[0] ? 0 : HDA_AMP_MUTE);
-	change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-					   HDA_AMP_MUTE,
-					   valp[1] ? 0 : HDA_AMP_MUTE);
 	if (change)
 		alc268_acer_automute(codec, 0);
 	return change;

sound/pci/hda/patch_sigmatel.c

@@ -4066,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
 	jack->nid = nid;
 	jack->type = type;
 
-	sprintf(name, "%s at %s %s Jack",
+	snprintf(name, sizeof(name), "%s at %s %s Jack",
 		snd_hda_get_jack_type(def_conf),
 		snd_hda_get_jack_connectivity(def_conf),
 		snd_hda_get_jack_location(def_conf));

sound/soc/codecs/tlv320aic3x.c

@@ -767,6 +767,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
 	u16 pll_d = 1;
+	u8 reg;
 
 	/* select data word length */
 	data =
@@ -801,8 +802,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 		pll_q &= 0xf;
 		aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
 		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
-	} else
+		/* disable PLL if it is bypassed */
+		reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+		aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE);
+
+	} else {
 		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+		/* enable PLL when it is used */
+		reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+		aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE);
+	}
 
 	/* Route Left DAC to left channel input and
 	 * right DAC to right channel input */

sound/usb/Kconfig

@@ -48,6 +48,7 @@ config SND_USB_CAIAQ
 	    * Native Instruments Kore Controller
 	    * Native Instruments Kore Controller 2
 	    * Native Instruments Audio Kontrol 1
+	    * Native Instruments Audio 2 DJ
 	    * Native Instruments Audio 4 DJ
 	    * Native Instruments Audio 8 DJ
 	    * Native Instruments Guitar Rig Session I/O

sound/usb/caiaq/audio.c

@@ -646,6 +646,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
 		dev->samplerates |= SNDRV_PCM_RATE_192000;
 		/* fall thru */
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
 		dev->samplerates |= SNDRV_PCM_RATE_88200;

sound/usb/caiaq/device.c

@@ -35,13 +35,14 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
 			 "{Native Instruments, Kore Controller},"
 			 "{Native Instruments, Kore Controller 2},"
 			 "{Native Instruments, Audio Kontrol 1},"
+			 "{Native Instruments, Audio 2 DJ},"
 			 "{Native Instruments, Audio 4 DJ},"
 			 "{Native Instruments, Audio 8 DJ},"
 			 "{Native Instruments, Session I/O},"
@@ -121,6 +122,11 @@ static struct usb_device_id snd_usb_id_table[] = {
 		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
 		.idProduct =    USB_PID_AUDIO4DJ
 	},
+	{
+		.match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
+		.idProduct =    USB_PID_AUDIO2DJ
+	},
 	{ /* terminator */ }
 };
 

sound/usb/caiaq/device.h

@@ -10,6 +10,7 @@
 #define USB_PID_KORECONTROLLER	0x4711
 #define USB_PID_KORECONTROLLER2	0x4712
 #define USB_PID_AK1		0x0815
+#define USB_PID_AUDIO2DJ	0x041c
 #define USB_PID_AUDIO4DJ	0x0839
 #define USB_PID_AUDIO8DJ	0x1978
 #define USB_PID_SESSIONIO	0x1915

sound/usb/usbmixer.c

@@ -990,20 +990,35 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
 		break;
 	}
 
-	/* quirk for UDA1321/N101 */
+	/* volume control quirks */
-	/* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
-	/* is not very clear from datasheets */
-	/* I hope that the min value is -15360 for newer firmware --jk */
 	switch (state->chip->usb_id) {
 	case USB_ID(0x0471, 0x0101):
 	case USB_ID(0x0471, 0x0104):
 	case USB_ID(0x0471, 0x0105):
 	case USB_ID(0x0672, 0x1041):
+	/* quirk for UDA1321/N101.
+	 * note that detection between firmware 2.1.1.7 (N101)
+	 * and later 2.1.1.21 is not very clear from datasheets.
+	 * I hope that the min value is -15360 for newer firmware --jk
+	 */
 		if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
 		    cval->min == -15616) {
-			snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n");
+			snd_printk(KERN_INFO
+				 "set volume quirk for UDA1321/N101 chip\n");
 			cval->max = -256;
 		}
+		break;
+
+	case USB_ID(0x046d, 0x09a4):
+		if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+			snd_printk(KERN_INFO
+				"set volume quirk for QuickCam E3500\n");
+			cval->min = 6080;
+			cval->max = 8768;
+			cval->res = 192;
+		}
+		break;
+
 	}
 
 	snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",