Merge branch 'topic/misc' into for-linus

52 files changed, 2690 insertions, 1111 deletions

diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c
index a351dd0a09c7..2b50cbe6aca9 100644
--- a/sound/ac97_bus.c
+++ b/sound/ac97_bus.c
@@ -19,8 +19,8 @@
 
 /*
  * Let drivers decide whether they want to support given codec from their
- * probe method.  Drivers have direct access to the struct snd_ac97 structure and may
- * decide based on the id field amongst other things.
+ * probe method. Drivers have direct access to the struct snd_ac97
+ * structure and may  decide based on the id field amongst other things.
  */
 static int ac97_bus_match(struct device *dev, struct device_driver *drv)
 {
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index 91852e49910e..3687a6cc9881 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -1114,7 +1114,6 @@ static int onyx_i2c_remove(struct i2c_client *client)
 	of_node_put(onyx->codec.node);
 	if (onyx->codec_info)
 		kfree(onyx->codec_info);
-	i2c_set_clientdata(client, onyx);
 	kfree(onyx);
 	return 0;
 }
diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c
index de8e03afa97b..faa317490545 100644
--- a/sound/aoa/core/gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -287,10 +287,9 @@ static void ftr_gpio_exit(struct gpio_runtime *rt)
 		free_irq(linein_detect_irq, &rt->line_in_notify);
 	if (rt->line_out_notify.gpio_private)
 		free_irq(lineout_detect_irq, &rt->line_out_notify);
-	cancel_delayed_work(&rt->headphone_notify.work);
-	cancel_delayed_work(&rt->line_in_notify.work);
-	cancel_delayed_work(&rt->line_out_notify.work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
 	mutex_destroy(&rt->headphone_notify.mutex);
 	mutex_destroy(&rt->line_in_notify.mutex);
 	mutex_destroy(&rt->line_out_notify.mutex);
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 7e267c9379bc..c8d8a1a6f964 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -107,10 +107,9 @@ static void pmf_gpio_exit(struct gpio_runtime *rt)
 
 	/* make sure no work is pending before freeing
 	 * all things */
-	cancel_delayed_work(&rt->headphone_notify.work);
-	cancel_delayed_work(&rt->line_in_notify.work);
-	cancel_delayed_work(&rt->line_out_notify.work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
 
 	mutex_destroy(&rt->headphone_notify.mutex);
 	mutex_destroy(&rt->line_in_notify.mutex);
diff --git a/sound/core/control.c b/sound/core/control.c
index 45a818002d99..9ce00ed20fba 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1488,7 +1488,7 @@ int snd_ctl_create(struct snd_card *card)
 }
 
 /*
- * Frequently used control callbacks
+ * Frequently used control callbacks/helpers
  */
 int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_info *uinfo)
@@ -1513,3 +1513,29 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
 }
 
 EXPORT_SYMBOL(snd_ctl_boolean_stereo_info);
+
+/**
+ * snd_ctl_enum_info - fills the info structure for an enumerated control
+ * @info: the structure to be filled
+ * @channels: the number of the control's channels; often one
+ * @items: the number of control values; also the size of @names
+ * @names: an array containing the names of all control values
+ *
+ * Sets all required fields in @info to their appropriate values.
+ * If the control's accessibility is not the default (readable and writable),
+ * the caller has to fill @info->access.
+ */
+int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
+		      unsigned int items, const char *const names[])
+{
+	info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	info->count = channels;
+	info->value.enumerated.items = items;
+	if (info->value.enumerated.item >= items)
+		info->value.enumerated.item = items - 1;
+	strlcpy(info->value.enumerated.name,
+		names[info->value.enumerated.item],
+		sizeof(info->value.enumerated.name));
+	return 0;
+}
+EXPORT_SYMBOL(snd_ctl_enum_info);
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index b753ec661fcf..a2e4eb324699 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -453,8 +453,10 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm,
 	} else {
 		*params = *save;
 		max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir);
-		if (max < 0)
+		if (max < 0) {
+			kfree(save);
 			return max;
+		}
 		last = 1;
 	}
  _end:
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 11446a1506da..a82e3756a72d 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -373,6 +373,27 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			   (unsigned long)new_hw_ptr,
 			   (unsigned long)runtime->hw_ptr_base);
 	}
+
+	if (runtime->no_period_wakeup) {
+		/*
+		 * Without regular period interrupts, we have to check
+		 * the elapsed time to detect xruns.
+		 */
+		jdelta = jiffies - runtime->hw_ptr_jiffies;
+		if (jdelta < runtime->hw_ptr_buffer_jiffies / 2)
+			goto no_delta_check;
+		hdelta = jdelta - delta * HZ / runtime->rate;
+		while (hdelta > runtime->hw_ptr_buffer_jiffies / 2 + 1) {
+			delta += runtime->buffer_size;
+			hw_base += runtime->buffer_size;
+			if (hw_base >= runtime->boundary)
+				hw_base = 0;
+			new_hw_ptr = hw_base + pos;
+			hdelta -= runtime->hw_ptr_buffer_jiffies;
+		}
+		goto no_delta_check;
+	}
+
 	/* something must be really wrong */
 	if (delta >= runtime->buffer_size + runtime->period_size) {
 		hw_ptr_error(substream,
@@ -442,6 +463,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			     (long)old_hw_ptr);
 	}
 
+ no_delta_check:
 	if (runtime->status->hw_ptr == new_hw_ptr)
 		return 0;
 
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index e82c1f97d99e..0db714e87a80 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -422,6 +422,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 	runtime->info = params->info;
 	runtime->rate_num = params->rate_num;
 	runtime->rate_den = params->rate_den;
+	runtime->no_period_wakeup =
+			(params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) &&
+			(params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP);
 
 	bits = snd_pcm_format_physical_width(runtime->format);
 	runtime->sample_bits = bits;
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index bf09a5ad1865..119fddb6fc99 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -32,6 +32,7 @@
 #include "seq_timer.h"
 #include "seq_system.h"
 #include "seq_info.h"
+#include <sound/minors.h>
 #include <sound/seq_device.h>
 
 #if defined(CONFIG_SND_SEQ_DUMMY_MODULE)
@@ -73,6 +74,9 @@ MODULE_PARM_DESC(seq_default_timer_subdevice, "The default timer subdevice numbe
 module_param(seq_default_timer_resolution, int, 0644);
 MODULE_PARM_DESC(seq_default_timer_resolution, "The default timer resolution in Hz.");
 
+MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_SEQUENCER);
+MODULE_ALIAS("devname:snd/seq");
+
 /*
  *  INIT PART
  */
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 66691fe437e6..1c7a3efe1778 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -188,14 +188,22 @@ static const struct file_operations snd_fops =
 };
 
 #ifdef CONFIG_SND_DYNAMIC_MINORS
-static int snd_find_free_minor(void)
+static int snd_find_free_minor(int type)
 {
 	int minor;
 
+	/* static minors for module auto loading */
+	if (type == SNDRV_DEVICE_TYPE_SEQUENCER)
+		return SNDRV_MINOR_SEQUENCER;
+	if (type == SNDRV_DEVICE_TYPE_TIMER)
+		return SNDRV_MINOR_TIMER;
+
 	for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor) {
-		/* skip minors still used statically for autoloading devices */
-		if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL ||
-		    minor == SNDRV_MINOR_SEQUENCER)
+		/* skip static minors still used for module auto loading */
+		if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL)
+			continue;
+		if (minor == SNDRV_MINOR_SEQUENCER ||
+		    minor == SNDRV_MINOR_TIMER)
 			continue;
 		if (!snd_minors[minor])
 			return minor;
@@ -269,7 +277,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
 	preg->private_data = private_data;
 	mutex_lock(&sound_mutex);
 #ifdef CONFIG_SND_DYNAMIC_MINORS
-	minor = snd_find_free_minor();
+	minor = snd_find_free_minor(type);
 #else
 	minor = snd_kernel_minor(type, card, dev);
 	if (minor >= 0 && snd_minors[minor])
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 13afb60999b9..ed016329e911 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -34,8 +34,8 @@
 #include <sound/initval.h>
 #include <linux/kmod.h>
 
-#if defined(CONFIG_SND_HPET) || defined(CONFIG_SND_HPET_MODULE)
-#define DEFAULT_TIMER_LIMIT 3
+#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE)
+#define DEFAULT_TIMER_LIMIT 4
 #elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE)
 #define DEFAULT_TIMER_LIMIT 2
 #else
@@ -52,6 +52,9 @@ MODULE_PARM_DESC(timer_limit, "Maximum global timers in system.");
 module_param(timer_tstamp_monotonic, int, 0444);
 MODULE_PARM_DESC(timer_tstamp_monotonic, "Use posix monotonic clock source for timestamps (default).");
 
+MODULE_ALIAS_CHARDEV(CONFIG_SND_MAJOR, SNDRV_MINOR_TIMER);
+MODULE_ALIAS("devname:snd/timer");
+
 struct snd_timer_user {
 	struct snd_timer_instance *timeri;
 	int tread;		/* enhanced read with timestamps and events */
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index a1282c1c0591..5cfcb908c430 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1143,8 +1143,8 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 					     (resource->start) + 1);
 	if (ml403_ac97cr->port == NULL) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
-			   "unable to remap memory region (%x to %x)\n",
-			   resource->start, resource->end);
+			   "unable to remap memory region (%pR)\n",
+			   resource);
 		snd_ml403_ac97cr_free(ml403_ac97cr);
 		return -EBUSY;
 	}
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index 971a84a4fa77..c424d329f806 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -57,8 +57,7 @@ static void snd_ak4113_free(struct ak4113 *chip)
 {
 	chip->init = 1;	/* don't schedule new work */
 	mb();
-	cancel_delayed_work(&chip->work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&chip->work);
 	kfree(chip);
 }
 
@@ -141,7 +140,7 @@ void snd_ak4113_reinit(struct ak4113 *chip)
 {
 	chip->init = 1;
 	mb();
-	flush_scheduled_work();
+	flush_delayed_work_sync(&chip->work);
 	ak4113_init_regs(chip);
 	/* bring up statistics / event queing */
 	chip->init = 0;
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
index 0341451f814c..d9fb537b0b94 100644
--- a/sound/i2c/other/ak4114.c
+++ b/sound/i2c/other/ak4114.c
@@ -67,8 +67,7 @@ static void snd_ak4114_free(struct ak4114 *chip)
 {
 	chip->init = 1;	/* don't schedule new work */
 	mb();
-	cancel_delayed_work(&chip->work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&chip->work);
 	kfree(chip);
 }
 
@@ -154,7 +153,7 @@ void snd_ak4114_reinit(struct ak4114 *chip)
 {
 	chip->init = 1;
 	mb();
-	flush_scheduled_work();
+	flush_delayed_work_sync(&chip->work);
 	ak4114_init_regs(chip);
 	/* bring up statistics / event queing */
 	chip->init = 0;
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 12e34653b8a8..9823d59d7ad7 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -209,7 +209,7 @@ config SND_OXYGEN_LIB
         tristate
 
 config SND_OXYGEN
-	tristate "C-Media 8788 (Oxygen)"
+	tristate "C-Media 8786, 8787, 8788 (Oxygen)"
 	select SND_OXYGEN_LIB
 	select SND_PCM
 	select SND_MPU401_UART
@@ -217,13 +217,18 @@ config SND_OXYGEN
 	  Say Y here to include support for sound cards based on the
 	  C-Media CMI8788 (Oxygen HD Audio) chip:
 	   * Asound A-8788
+	   * Asus Xonar DG
 	   * AuzenTech X-Meridian
+	   * AuzenTech X-Meridian 2G
 	   * Bgears b-Enspirer
 	   * Club3D Theatron DTS
 	   * HT-Omega Claro (plus)
 	   * HT-Omega Claro halo (XT)
+	   * Kuroutoshikou CMI8787-HG2PCI
 	   * Razer Barracuda AC-1
 	   * Sondigo Inferno
+	   * TempoTec/MediaTek HiFier Fantasia
+	   * TempoTec/MediaTek HiFier Serenade
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-oxygen.
@@ -578,18 +583,6 @@ config SND_HDSPM
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-hdspm.
 
-config SND_HIFIER
-	tristate "TempoTec HiFier Fantasia"
-	select SND_OXYGEN_LIB
-	select SND_PCM
-	select SND_MPU401_UART
-	help
-	  Say Y here to include support for the MediaTek/TempoTec HiFier
-	  Fantasia sound card.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-hifier.
-
 config SND_ICE1712
 	tristate "ICEnsemble ICE1712 (Envy24)"
 	select SND_MPU401_UART
@@ -826,8 +819,8 @@ config SND_VIRTUOSO
 	  Say Y here to include support for sound cards based on the
 	  Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS,
 	  Essence ST (Deluxe), and Essence STX.
-	  Support for the HDAV1.3 (Deluxe) is incomplete; for the
-	  HDAV1.3 Slim and Xense, missing.
+	  Support for the HDAV1.3 (Deluxe) and HDAV1.3 Slim is experimental;
+	  for the Xense, missing.
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-virtuoso.
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index a7630e9edf8a..0fc614ce16c1 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1014,8 +1014,7 @@ static int snd_ac97_free(struct snd_ac97 *ac97)
 {
 	if (ac97) {
 #ifdef CONFIG_SND_AC97_POWER_SAVE
-		cancel_delayed_work(&ac97->power_work);
-		flush_scheduled_work();
+		cancel_delayed_work_sync(&ac97->power_work);
 #endif
 		snd_ac97_proc_done(ac97);
 		if (ac97->bus)
@@ -2456,8 +2455,7 @@ void snd_ac97_suspend(struct snd_ac97 *ac97)
 	if (ac97->build_ops->suspend)
 		ac97->build_ops->suspend(ac97);
 #ifdef CONFIG_SND_AC97_POWER_SAVE
-	cancel_delayed_work(&ac97->power_work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&ac97->power_work);
 #endif
 	snd_ac97_powerdown(ac97);
 }
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 2f3cacbd5528..6117595fc075 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1,6 +1,6 @@
 /*
  *  azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168).
- *  Copyright (C) 2002, 2005 - 2009 by Andreas Mohr <andi AT lisas.de>
+ *  Copyright (C) 2002, 2005 - 2010 by Andreas Mohr <andi AT lisas.de>
  *
  *  Framework borrowed from Bart Hartgers's als4000.c.
  *  Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801),
@@ -175,6 +175,7 @@
 
 #include <asm/io.h>
 #include <linux/init.h>
+#include <linux/bug.h> /* WARN_ONCE */
 #include <linux/pci.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
@@ -201,14 +202,15 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 
 /* === Debug settings ===
   Further diagnostic functionality than the settings below
-  does not need to be provided, since one can easily write a bash script
+  does not need to be provided, since one can easily write a POSIX shell script
   to dump the card's I/O ports (those listed in lspci -v -v):
-  function dump()
+  dump()
   {
     local descr=$1; local addr=$2; local count=$3
 
     echo "${descr}: ${count} @ ${addr}:"
-    dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C
+    dd if=/dev/port skip=`printf %d ${addr}` count=${count} bs=1 \
+      2>/dev/null| hexdump -C
   }
   and then use something like
   "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8",
@@ -216,14 +218,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
   possibly within a "while true; do ... sleep 1; done" loop.
   Tweaking ports could be done using
   VALSTRING="`printf "%02x" $value`"
-  printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null
+  printf "\x""$VALSTRING"|dd of=/dev/port seek=`printf %d ${addr}` bs=1 \
+    2>/dev/null
 */
 
 #define DEBUG_MISC	0
 #define DEBUG_CALLS	0
 #define DEBUG_MIXER	0
 #define DEBUG_CODEC	0
-#define DEBUG_IO	0
 #define DEBUG_TIMER	0
 #define DEBUG_GAME	0
 #define DEBUG_PM	0
@@ -291,19 +293,23 @@ static int seqtimer_scaling = 128;
 module_param(seqtimer_scaling, int, 0444);
 MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128.");
 
-struct snd_azf3328_codec_data {
-	unsigned long io_base;
-	struct snd_pcm_substream *substream;
-	bool running;
-	const char *name;
-};
-
 enum snd_azf3328_codec_type {
+  /* warning: fixed indices (also used for bitmask checks!) */
   AZF_CODEC_PLAYBACK = 0,
   AZF_CODEC_CAPTURE = 1,
   AZF_CODEC_I2S_OUT = 2,
 };
 
+struct snd_azf3328_codec_data {
+	unsigned long io_base; /* keep first! (avoid offset calc) */
+	unsigned int dma_base; /* helper to avoid an indirection in hotpath */
+	spinlock_t *lock; /* TODO: convert to our own per-codec lock member */
+	struct snd_pcm_substream *substream;
+	bool running;
+	enum snd_azf3328_codec_type type;
+	const char *name;
+};
+
 struct snd_azf3328 {
 	/* often-used fields towards beginning, then grouped */
 
@@ -362,6 +368,9 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids);
 static int
 snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set)
 {
+	/* Well, strictly spoken, the inb/outb sequence isn't atomic
+	   and would need locking. However we currently don't care
+	   since it potentially complicates matters. */
 	u8 prev = inb(reg), new;
 
 	new = (do_set) ? (prev|mask) : (prev & ~mask);
@@ -413,6 +422,21 @@ snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec,
 	outl(value, codec->io_base + reg);
 }
 
+static inline void
+snd_azf3328_codec_outl_multi(const struct snd_azf3328_codec_data *codec,
+			     unsigned reg, const void *buffer, int count
+)
+{
+	unsigned long addr = codec->io_base + reg;
+	if (count) {
+		const u32 *buf = buffer;
+		do {
+			outl(*buf++, addr);
+			addr += 4;
+		} while (--count);
+	}
+}
+
 static inline u32
 snd_azf3328_codec_inl(const struct snd_azf3328_codec_data *codec, unsigned reg)
 {
@@ -943,38 +967,43 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream)
 }
 
 static void
-snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
-			       enum snd_azf3328_codec_type codec_type,
+snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
 			       enum azf_freq_t bitrate,
 			       unsigned int format_width,
 			       unsigned int channels
 )
 {
 	unsigned long flags;
-	const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type];
 	u16 val = 0xff00;
+	u8 freq = 0;
 
 	snd_azf3328_dbgcallenter();
 	switch (bitrate) {
-	case AZF_FREQ_4000:  val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
-	case AZF_FREQ_4800:  val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
-	case AZF_FREQ_5512:
-		/* the AZF3328 names it "5510" for some strange reason */
-			     val |= SOUNDFORMAT_FREQ_5510; break;
-	case AZF_FREQ_6620:  val |= SOUNDFORMAT_FREQ_6620; break;
-	case AZF_FREQ_8000:  val |= SOUNDFORMAT_FREQ_8000; break;
-	case AZF_FREQ_9600:  val |= SOUNDFORMAT_FREQ_9600; break;
-	case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break;
-	case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
-	case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break;
-	case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break;
-	case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break;
+#define AZF_FMT_XLATE(in_freq, out_bits) \
+	do { \
+		case AZF_FREQ_ ## in_freq: \
+			freq = SOUNDFORMAT_FREQ_ ## out_bits; \
+			break; \
+	} while (0);
+	AZF_FMT_XLATE(4000, SUSPECTED_4000)
+	AZF_FMT_XLATE(4800, SUSPECTED_4800)
+	/* the AZF3328 names it "5510" for some strange reason: */
+	AZF_FMT_XLATE(5512, 5510)
+	AZF_FMT_XLATE(6620, 6620)
+	AZF_FMT_XLATE(8000, 8000)
+	AZF_FMT_XLATE(9600, 9600)
+	AZF_FMT_XLATE(11025, 11025)
+	AZF_FMT_XLATE(13240, SUSPECTED_13240)
+	AZF_FMT_XLATE(16000, 16000)
+	AZF_FMT_XLATE(22050, 22050)
+	AZF_FMT_XLATE(32000, 32000)
 	default:
 		snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
 		/* fall-through */
-	case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break;
-	case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break;
-	case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
+	AZF_FMT_XLATE(44100, 44100)
+	AZF_FMT_XLATE(48000, 48000)
+	AZF_FMT_XLATE(66200, SUSPECTED_66200)
+#undef AZF_FMT_XLATE
 	}
 	/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
 	/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
@@ -986,13 +1015,15 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
 	/* val = 0xff0d; 41m23.135s (5523,600Hz; -> 5512Hz???) */
 	/* val = 0xff0e; 28m30.777s (8017Hz; -> 8000Hz???) */
 
+	val |= freq;
+
 	if (channels == 2)
 		val |= SOUNDFORMAT_FLAG_2CHANNELS;
 
 	if (format_width == 16)
 		val |= SOUNDFORMAT_FLAG_16BIT;
 
-	spin_lock_irqsave(&chip->reg_lock, flags);
+	spin_lock_irqsave(codec->lock, flags);
 
 	/* set bitrate/format */
 	snd_azf3328_codec_outw(codec, IDX_IO_CODEC_SOUNDFORMAT, val);
@@ -1004,7 +1035,8 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
 	 * (FIXME: yes, it works, but what exactly am I doing here?? :)
 	 * FIXME: does this have some side effects for full-duplex
 	 * or other dramatic side effects? */
-	if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */
+	/* do it for non-capture codecs only */
+	if (codec->type != AZF_CODEC_CAPTURE)
 		snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS,
 			snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) |
 			DMA_RUN_SOMETHING1