Merge branch 'for-linus' into for-next

Merging the HD-audio fixes back to base devel branch for further
working on it.
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
index c949abc..c3495be 100644
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt
@@ -18,6 +18,7 @@
   * Headphones
   * Speakers
   * Mic Jack
+  * Int Mic
 
 - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
   connected to the CODEC.
diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
new file mode 100644
index 0000000..0b191a2
--- /dev/null
+++ b/Documentation/sound/alsa/timestamping.txt
@@ -0,0 +1,200 @@
+The ALSA API can provide two different system timestamps:
+
+- Trigger_tstamp is the system time snapshot taken when the .trigger
+callback is invoked. This snapshot is taken by the ALSA core in the
+general case, but specific hardware may have synchronization
+capabilities or conversely may only be able to provide a correct
+estimate with a delay. In the latter two cases, the low-level driver
+is responsible for updating the trigger_tstamp at the most appropriate
+and precise moment. Applications should not rely solely on the first
+trigger_tstamp but update their internal calculations if the driver
+provides a refined estimate with a delay.
+
+- tstamp is the current system timestamp updated during the last
+event or application query.
+The difference (tstamp - trigger_tstamp) defines the elapsed time.
+
+The ALSA API provides reports two basic pieces of information, avail
+and delay, which combined with the trigger and current system
+timestamps allow for applications to keep track of the 'fullness' of
+the ring buffer and the amount of queued samples.
+
+The use of these different pointers and time information depends on
+the application needs:
+
+- 'avail' reports how much can be written in the ring buffer
+- 'delay' reports the time it will take to hear a new sample after all
+queued samples have been played out.
+
+When timestamps are enabled, the avail/delay information is reported
+along with a snapshot of system time. Applications can select from
+CLOCK_REALTIME (NTP corrections including going backwards),
+CLOCK_MONOTONIC (NTP corrections but never going backwards),
+CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
+dynamically with sw_params
+
+
+The ALSA API also provide an audio_tstamp which reflects the passage
+of time as measured by different components of audio hardware.  In
+ascii-art, this could be represented as follows (for the playback
+case):
+
+
+--------------------------------------------------------------> time
+  ^               ^              ^                ^           ^
+  |               |              |                |           |
+ analog         link            dma              app       FullBuffer
+ time           time           time              time        time
+  |               |              |                |           |
+  |< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
+  |<----------------- delay---------------------->|           |
+			         |<----ring buffer length---->|
+
+The analog time is taken at the last stage of the playback, as close
+as possible to the actual transducer
+
+The link time is taken at the output of the SOC/chipset as the samples
+are pushed on a link. The link time can be directly measured if
+supported in hardware by sample counters or wallclocks (e.g. with
+HDAudio 24MHz or PTP clock for networked solutions) or indirectly
+estimated (e.g. with the frame counter in USB).
+
+The DMA time is measured using counters - typically the least reliable
+of all measurements due to the bursty natured of DMA transfers.
+
+The app time corresponds to the time tracked by an application after
+writing in the ring buffer.
+
+The application can query what the hardware supports, define which
+audio time it wants reported by selecting the relevant settings in
+audio_tstamp_config fields, get an estimate of the timestamp
+accuracy. It can also request the delay-to-analog be included in the
+measurement. Direct access to the link time is very interesting on
+platforms that provide an embedded DSP; measuring directly the link
+time with dedicated hardware, possibly synchronized with system time,
+removes the need to keep track of internal DSP processing times and
+latency.
+
+In case the application requests an audio tstamp that is not supported
+in hardware/low-level driver, the type is overridden as DEFAULT and the
+timestamp will report the DMA time based on the hw_pointer value.
+
+For backwards compatibility with previous implementations that did not
+provide timestamp selection, with a zero-valued COMPAT timestamp type
+the results will default to the HDAudio wall clock for playback
+streams and to the DMA time (hw_ptr) in all other cases.
+
+The audio timestamp accuracy can be returned to user-space, so that
+appropriate decisions are made:
+
+- for dma time (default), the granularity of the transfers can be
+  inferred from the steps between updates and in turn provide
+  information on how much the application pointer can be rewound
+  safely.
+
+- the link time can be used to track long-term drifts between audio
+  and system time using the (tstamp-trigger_tstamp)/audio_tstamp
+  ratio, the precision helps define how much smoothing/low-pass
+  filtering is required. The link time can be either reset on startup
+  or reported as is (the latter being useful to compare progress of
+  different streams - but may require the wallclock to be always
+  running and not wrap-around during idle periods). If supported in
+  hardware, the absolute link time could also be used to define a
+  precise start time (patches WIP)
+
+- including the delay in the audio timestamp may
+  counter-intuitively not increase the precision of timestamps, e.g. if a
+  codec includes variable-latency DSP processing or a chain of
+  hardware components the delay is typically not known with precision.
+
+The accuracy is reported in nanosecond units (using an unsigned 32-bit
+word), which gives a max precision of 4.29s, more than enough for
+audio applications...
+
+Due to the varied nature of timestamping needs, even for a single
+application, the audio_tstamp_config can be changed dynamically. In
+the STATUS ioctl, the parameters are read-only and do not allow for
+any application selection. To work around this limitation without
+impacting legacy applications, a new STATUS_EXT ioctl is introduced
+with read/write parameters. ALSA-lib will be modified to make use of
+STATUS_EXT and effectively deprecate STATUS.
+
+The ALSA API only allows for a single audio timestamp to be reported
+at a time. This is a conscious design decision, reading the audio
+timestamps from hardware registers or from IPC takes time, the more
+timestamps are read the more imprecise the combined measurements
+are. To avoid any interpretation issues, a single (system, audio)
+timestamp is reported. Applications that need different timestamps
+will be required to issue multiple queries and perform an
+interpolation of the results
+
+In some hardware-specific configuration, the system timestamp is
+latched by a low-level audio subsytem, and the information provided
+back to the driver. Due to potential delays in the communication with
+the hardware, there is a risk of misalignment with the avail and delay
+information. To make sure applications are not confused, a
+driver_timestamp field is added in the snd_pcm_status structure; this
+timestamp shows when the information is put together by the driver
+before returning from the STATUS and STATUS_EXT ioctl. in most cases
+this driver_timestamp will be identical to the regular system tstamp.
+
+Examples of typestamping with HDaudio:
+
+1. DMA timestamp, no compensation for DMA+analog delay
+$ ./audio_time  -p --ts_type=1
+playback: systime: 341121338 nsec, audio time 342000000 nsec, 	systime delta -878662
+playback: systime: 426236663 nsec, audio time 427187500 nsec, 	systime delta -950837
+playback: systime: 597080580 nsec, audio time 598000000 nsec, 	systime delta -919420
+playback: systime: 682059782 nsec, audio time 683020833 nsec, 	systime delta -961051
+playback: systime: 852896415 nsec, audio time 853854166 nsec, 	systime delta -957751
+playback: systime: 937903344 nsec, audio time 938854166 nsec, 	systime delta -950822
+
+2. DMA timestamp, compensation for DMA+analog delay
+$ ./audio_time  -p --ts_type=1 -d
+playback: systime: 341053347 nsec, audio time 341062500 nsec, 	systime delta -9153
+playback: systime: 426072447 nsec, audio time 426062500 nsec, 	systime delta 9947
+playback: systime: 596899518 nsec, audio time 596895833 nsec, 	systime delta 3685
+playback: systime: 681915317 nsec, audio time 681916666 nsec, 	systime delta -1349
+playback: systime: 852741306 nsec, audio time 852750000 nsec, 	systime delta -8694
+
+3. link timestamp, compensation for DMA+analog delay
+$ ./audio_time  -p --ts_type=2 -d
+playback: systime: 341060004 nsec, audio time 341062791 nsec, 	systime delta -2787
+playback: systime: 426242074 nsec, audio time 426244875 nsec, 	systime delta -2801
+playback: systime: 597080992 nsec, audio time 597084583 nsec, 	systime delta -3591
+playback: systime: 682084512 nsec, audio time 682088291 nsec, 	systime delta -3779
+playback: systime: 852936229 nsec, audio time 852940916 nsec, 	systime delta -4687
+playback: systime: 938107562 nsec, audio time 938112708 nsec, 	systime delta -5146
+
+Example 1 shows that the timestamp at the DMA level is close to 1ms
+ahead of the actual playback time (as a side time this sort of
+measurement can help define rewind safeguards). Compensating for the
+DMA-link delay in example 2 helps remove the hardware buffering abut
+the information is still very jittery, with up to one sample of
+error. In example 3 where the timestamps are measured with the link
+wallclock, the timestamps show a monotonic behavior and a lower
+dispersion.
+
+Example 3 and 4 are with USB audio class. Example 3 shows a high
+offset between audio time and system time due to buffering. Example 4
+shows how compensating for the delay exposes a 1ms accuracy (due to
+the use of the frame counter by the driver)
+
+Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
+$ ./audio_time -p -Dhw:1 -t1
+playback: systime: 120174019 nsec, audio time 125000000 nsec, 	systime delta -4825981
+playback: systime: 245041136 nsec, audio time 250000000 nsec, 	systime delta -4958864
+playback: systime: 370106088 nsec, audio time 375000000 nsec, 	systime delta -4893912
+playback: systime: 495040065 nsec, audio time 500000000 nsec, 	systime delta -4959935
+playback: systime: 620038179 nsec, audio time 625000000 nsec, 	systime delta -4961821
+playback: systime: 745087741 nsec, audio time 750000000 nsec, 	systime delta -4912259
+playback: systime: 870037336 nsec, audio time 875000000 nsec, 	systime delta -4962664
+
+Example 4: DMA timestamp, compensation for delay, delay of ~1ms
+$ ./audio_time -p -Dhw:1 -t1 -d
+playback: systime: 120190520 nsec, audio time 120000000 nsec, 	systime delta 190520
+playback: systime: 245036740 nsec, audio time 244000000 nsec, 	systime delta 1036740
+playback: systime: 370034081 nsec, audio time 369000000 nsec, 	systime delta 1034081
+playback: systime: 495159907 nsec, audio time 494000000 nsec, 	systime delta 1159907
+playback: systime: 620098824 nsec, audio time 619000000 nsec, 	systime delta 1098824
+playback: systime: 745031847 nsec, audio time 744000000 nsec, 	systime delta 1031847
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index f48089d..fa1d055 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -70,7 +70,7 @@
  * @device: device pointer
  * @direction: stream direction, playback/recording
  * @metadata_set: metadata set flag, true when set
- * @next_track: has userspace signall next track transistion, true when set
+ * @next_track: has userspace signal next track transition, true when set
  * @private_data: pointer to DSP private data
  */
 struct snd_compr_stream {
@@ -95,7 +95,7 @@
  * and the stream properties
  * @get_params: retrieve the codec parameters, mandatory
  * @set_metadata: Set the metadata values for a stream
- * @get_metadata: retreives the requested metadata values from stream
+ * @get_metadata: retrieves the requested metadata values from stream
  * @trigger: Trigger operations like start, pause, resume, drain, stop.
  * This callback is mandatory
  * @pointer: Retrieve current h/w pointer information. Mandatory
diff --git a/include/sound/control.h b/include/sound/control.h
index 75f3054..95aad6d 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -227,7 +227,7 @@
  * Add a virtual slave control to the given master.
  * Unlike snd_ctl_add_slave(), the element added via this function
  * is supposed to have volatile values, and get callback is called
- * at each time quried from the master.
+ * at each time queried from the master.
  *
  * When the control peeks the hardware values directly and the value
  * can be changed by other means than the put callback of the element,
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index c0ddb7e..0cb7f3f 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -60,6 +60,9 @@
 
 struct snd_pcm_substream;
 
+struct snd_pcm_audio_tstamp_config; /* definitions further down */
+struct snd_pcm_audio_tstamp_report;
+
 struct snd_pcm_ops {
 	int (*open)(struct snd_pcm_substream *substream);
 	int (*close)(struct snd_pcm_substream *substream);
@@ -71,8 +74,10 @@
 	int (*prepare)(struct snd_pcm_substream *substream);
 	int (*trigger)(struct snd_pcm_substream *substream, int cmd);
 	snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream);
-	int (*wall_clock)(struct snd_pcm_substream *substream,
-			  struct timespec *audio_ts);
+	int (*get_time_info)(struct snd_pcm_substream *substream,
+			struct timespec *system_ts, struct timespec *audio_ts,
+			struct snd_pcm_audio_tstamp_config *audio_tstamp_config,
+			struct snd_pcm_audio_tstamp_report *audio_tstamp_report);
 	int (*copy)(struct snd_pcm_substream *substream, int channel,
 		    snd_pcm_uframes_t pos,
 		    void __user *buf, snd_pcm_uframes_t count);
@@ -281,6 +286,58 @@
 
 struct snd_pcm_hwptr_log;
 
+/*
+ * userspace-provided audio timestamp config to kernel,
+ * structure is for internal use only and filled with dedicated unpack routine
+ */
+struct snd_pcm_audio_tstamp_config {
+	/* 5 of max 16 bits used */
+	u32 type_requested:4;
+	u32 report_delay:1; /* add total delay to A/D or D/A */
+};
+
+static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data,
+						struct snd_pcm_audio_tstamp_config *config)
+{
+	config->type_requested = data & 0xF;
+	config->report_delay = (data >> 4) & 1;
+}
+
+/*
+ * kernel-provided audio timestamp report to user-space
+ * structure is for internal use only and read by dedicated pack routine
+ */
+struct snd_pcm_audio_tstamp_report {
+	/* 6 of max 16 bits used for bit-fields */
+
+	/* for backwards compatibility */
+	u32 valid:1;
+
+	/* actual type if hardware could not support requested timestamp */
+	u32 actual_type:4;
+
+	/* accuracy represented in ns units */
+	u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */
+	u32 accuracy; /* up to 4.29s, will be packed in separate field  */
+};
+
+static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy,
+						const struct snd_pcm_audio_tstamp_report *report)
+{
+	u32 tmp;
+
+	tmp = report->accuracy_report;
+	tmp <<= 4;
+	tmp |= report->actual_type;
+	tmp <<= 1;
+	tmp |= report->valid;
+
+	*data &= 0xffff; /* zero-clear MSBs */
+	*data |= (tmp << 16);
+	*accuracy = report->accuracy;
+}
+
+
 struct snd_pcm_runtime {
 	/* -- Status -- */
 	struct snd_pcm_substream *trigger_master;
@@ -361,6 +418,11 @@
 
 	struct snd_dma_buffer *dma_buffer_p;	/* allocated buffer */
 
+	/* -- audio timestamp config -- */
+	struct snd_pcm_audio_tstamp_config audio_tstamp_config;
+	struct snd_pcm_audio_tstamp_report audio_tstamp_report;
+	struct timespec driver_tstamp;
+
 #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
 	/* -- OSS things -- */
 	struct snd_pcm_oss_runtime oss;
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index 3c45f39..c704357 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -366,4 +366,11 @@
 	return snd_pcm_format_physical_width(params_format(p));
 }
 
+static inline void
+params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt)
+{
+	snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT),
+		(__force int)fmt);
+}
+
 #endif /* __SOUND_PCM_PARAMS_H */
diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h
index 2b5f24c..ddc0d50 100644
--- a/include/sound/seq_device.h
+++ b/include/sound/seq_device.h
@@ -25,29 +25,26 @@
  * registered device information
  */
 
-#define ID_LEN	32
-
-/* status flag */
-#define SNDRV_SEQ_DEVICE_FREE		0
-#define SNDRV_SEQ_DEVICE_REGISTERED	1
-
 struct snd_seq_device {
 	/* device info */
 	struct snd_card *card;	/* sound card */
 	int device;		/* device number */
-	char id[ID_LEN];	/* driver id */
+	const char *id;		/* driver id */
 	char name[80];		/* device name */
 	int argsize;		/* size of the argument */
 	void *driver_data;	/* private data for driver */
-	int status;		/* flag - read only */
 	void *private_data;	/* private data for the caller */
 	void (*private_free)(struct snd_seq_device *device);
-	struct list_head list;	/* link to next device */
+	struct device dev;
 };
 
+#define to_seq_dev(_dev) \
+	container_of(_dev, struct snd_seq_device, dev)
+
+/* sequencer driver */
 
 /* driver operators
- * init_device:
+ * probe:
  *	Initialize the device with given parameters.
  *	Typically,
  *		1. call snd_hwdep_new
@@ -55,25 +52,40 @@
  *		3. call snd_hwdep_register
  *		4. store the instance to dev->driver_data pointer.
  *		
- * free_device:
+ * remove:
  *	Release the private data.
  *	Typically, call snd_device_free(dev->card, dev->driver_data)
  */
-struct snd_seq_dev_ops {
-	int (*init_device)(struct snd_seq_device *dev);
-	int (*free_device)(struct snd_seq_device *dev);
+struct snd_seq_driver {
+	struct device_driver driver;
+	char *id;
+	int argsize;
 };
 
+#define to_seq_drv(_drv) \
+	container_of(_drv, struct snd_seq_driver, driver)
+
 /*
  * prototypes
  */
+#ifdef CONFIG_MODULES
 void snd_seq_device_load_drivers(void);
-int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result);
-int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize);
-int snd_seq_device_unregister_driver(char *id);
+#else
+#define snd_seq_device_load_drivers()
+#endif
+int snd_seq_device_new(struct snd_card *card, int device, const char *id,
+		       int argsize, struct snd_seq_device **result);
 
 #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device))
 
+int __must_check __snd_seq_driver_register(struct snd_seq_driver *drv,
+					   struct module *mod);
+#define snd_seq_driver_register(drv) \
+	__snd_seq_driver_register(drv, THIS_MODULE)
+void snd_seq_driver_unregister(struct snd_seq_driver *drv);
+
+#define module_snd_seq_driver(drv) \
+	module_driver(drv, snd_seq_driver_register, snd_seq_driver_unregister)
 
 /*
  * id strings for generic devices
diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h
index 18a2ac5..feb58d4 100644
--- a/include/sound/seq_kernel.h
+++ b/include/sound/seq_kernel.h
@@ -99,13 +99,9 @@
 int snd_seq_event_port_detach(int client, int port);
 
 #ifdef CONFIG_MODULES
-void snd_seq_autoload_lock(void);
-void snd_seq_autoload_unlock(void);
 void snd_seq_autoload_init(void);
-#define snd_seq_autoload_exit()	snd_seq_autoload_lock()
+void snd_seq_autoload_exit(void);
 #else
-#define snd_seq_autoload_lock()
-#define snd_seq_autoload_unlock()
 #define snd_seq_autoload_init()
 #define snd_seq_autoload_exit()
 #endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0d1ade1..b371aef 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -450,8 +450,10 @@
 		      struct snd_soc_dai *dai);
 
 /* Jack reporting */
-int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
-		     struct snd_soc_jack *jack);
+int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type,
+	struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins,
+	unsigned int num_pins);
+
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
 int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 			  struct snd_soc_jack_pin *pins);
@@ -659,7 +661,7 @@
 struct snd_soc_jack {
 	struct mutex mutex;
 	struct snd_jack *jack;
-	struct snd_soc_codec *codec;
+	struct snd_soc_card *card;
 	struct list_head pins;
 	int status;
 	struct blocking_notifier_head notifier;
@@ -954,6 +956,9 @@
 	unsigned int symmetric_channels:1;
 	unsigned int symmetric_samplebits:1;
 
+	/* Mark this pcm with non atomic ops */
+	bool nonatomic;
+
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int no_pcm:1;
 
@@ -1071,11 +1076,16 @@
 
 	/*
 	 * Card-specific routes and widgets.
+	 * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in.
 	 */
 	const struct snd_soc_dapm_widget *dapm_widgets;
 	int num_dapm_widgets;
 	const struct snd_soc_dapm_route *dapm_routes;
 	int num_dapm_routes;
+	const struct snd_soc_dapm_widget *of_dapm_widgets;
+	int num_of_dapm_widgets;
+	const struct snd_soc_dapm_route *of_dapm_routes;
+	int num_of_dapm_routes;
 	bool fully_routed;
 
 	struct work_struct deferred_resume_work;
@@ -1469,7 +1479,7 @@
 }
 
 /**
- * snd_soc_kcontrol_platform() - Returns the platform that registerd the control
+ * snd_soc_kcontrol_platform() - Returns the platform that registered the control
  * @kcontrol: The control for which to get the platform
  *
  * Note: This function will only work correctly if the control has been
diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h
index 09c8a00..5a5fa49 100644
--- a/include/uapi/sound/asequencer.h
+++ b/include/uapi/sound/asequencer.h
@@ -22,6 +22,7 @@
 #ifndef _UAPI__SOUND_ASEQUENCER_H
 #define _UAPI__SOUND_ASEQUENCER_H
 
+#include <sound/asound.h>
 
 /** version of the sequencer */
 #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1)
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 0e88e7a..46145a5 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -25,6 +25,9 @@
 
 #include <linux/types.h>
 
+#ifndef __KERNEL__
+#include <stdlib.h>
+#endif
 
 /*
  *  protocol version
@@ -140,7 +143,7 @@
  *                                                                           *
  *****************************************************************************/
 
-#define SNDRV_PCM_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 12)
+#define SNDRV_PCM_VERSION		SNDRV_PROTOCOL_VERSION(2, 0, 13)
 
 typedef unsigned long snd_pcm_uframes_t;
 typedef signed long snd_pcm_sframes_t;
@@ -267,10 +270,17 @@
 #define SNDRV_PCM_INFO_JOINT_DUPLEX	0x00200000	/* playback and capture stream are somewhat correlated */
 #define SNDRV_PCM_INFO_SYNC_START	0x00400000	/* pcm support some kind of sync go */
 #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP	0x00800000	/* period wakeup can be disabled */
-#define SNDRV_PCM_INFO_HAS_WALL_CLOCK   0x01000000      /* has audio wall clock for audio/system time sync */
+#define SNDRV_PCM_INFO_HAS_WALL_CLOCK   0x01000000      /* (Deprecated)has audio wall clock for audio/system time sync */
+#define SNDRV_PCM_INFO_HAS_LINK_ATIME              0x01000000  /* report hardware link audio time, reset on startup */
+#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME     0x02000000  /* report absolute hardware link audio time, not reset on startup */
+#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME    0x04000000  /* report estimated link audio time */
+#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000  /* report synchronized audio/system time */
+
 #define SNDRV_PCM_INFO_DRAIN_TRIGGER	0x40000000		/* internal kernel flag - trigger in drain */
 #define SNDRV_PCM_INFO_FIFO_IN_FRAMES	0x80000000	/* internal kernel flag - FIFO size is in frames */
 
+
+
 typedef int __bitwise snd_pcm_state_t;
 #define	SNDRV_PCM_STATE_OPEN		((__force snd_pcm_state_t) 0) /* stream is open */
 #define	SNDRV_PCM_STATE_SETUP		((__force snd_pcm_state_t) 1) /* stream has a setup */
@@ -408,6 +418,22 @@
 	unsigned int step;		/* samples distance in bits */
 };
 
+enum {
+	/*
+	 *  first definition for backwards compatibility only,
+	 *  maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else
+	 */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0,
+
+	/* timestamp definitions */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1,           /* DMA time, reported as per hw_ptr */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2,	           /* link time reported by sample or wallclock counter, reset on startup */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3,	   /* link time reported by sample or wallclock counter, not reset on startup */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4,    /* link time estimated indirectly */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */
+	SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED
+};
+
 struct snd_pcm_status {
 	snd_pcm_state_t state;		/* stream state */
 	struct timespec trigger_tstamp;	/* time when stream was started/stopped/paused */
@@ -419,9 +445,11 @@
 	snd_pcm_uframes_t avail_max;	/* max frames available on hw since last status */
 	snd_pcm_uframes_t overrange;	/* count of ADC (capture) overrange detections from last status */
 	snd_pcm_state_t suspended_state; /* suspended stream state */
-	__u32 reserved_alignment;	/* must be filled with zero */
-	struct timespec audio_tstamp;	/* from sample counter or wall clock */
-	unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */
+	__u32 audio_tstamp_data;	 /* needed for 64-bit alignment, used for configs/report to/from userspace */
+	struct timespec audio_tstamp;	/* sample counter, wall clock, PHC or on-demand sync'ed */
+	struct timespec driver_tstamp;	/* useful in case reference system tstamp is reported with delay */
+	__u32 audio_tstamp_accuracy;	/* in ns units, only valid if indicated in audio_tstamp_data */
+	unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */
 };
 
 struct snd_pcm_mmap_status {
@@ -534,6 +562,7 @@
 #define SNDRV_PCM_IOCTL_DELAY		_IOR('A', 0x21, snd_pcm_sframes_t)
 #define SNDRV_PCM_IOCTL_HWSYNC		_IO('A', 0x22)
 #define SNDRV_PCM_IOCTL_SYNC_PTR	_IOWR('A', 0x23, struct snd_pcm_sync_ptr)
+#define SNDRV_PCM_IOCTL_STATUS_EXT	_IOWR('A', 0x24, struct snd_pcm_status)
 #define SNDRV_PCM_IOCTL_CHANNEL_INFO	_IOR('A', 0x32, struct snd_pcm_channel_info)
 #define SNDRV_PCM_IOCTL_PREPARE		_IO('A', 0x40)
 #define SNDRV_PCM_IOCTL_RESET		_IO('A', 0x41)
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 22ed8cb..e00d8cb 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -75,7 +75,7 @@
 /**
  * struct snd_compr_avail - avail descriptor
  * @avail: Number of bytes available in ring buffer for writing/reading
- * @tstamp: timestamp infomation
+ * @tstamp: timestamp information
  */
 struct snd_compr_avail {
 	__u64 avail;
diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h
index d1bbaf7..ec1535b 100644
--- a/include/uapi/sound/emu10k1.h
+++ b/include/uapi/sound/emu10k1.h
@@ -23,8 +23,7 @@
 #define _UAPI__SOUND_EMU10K1_H
 
 #include <linux/types.h>
-
-
+#include <sound/asound.h>
 
 /*
  * ---- FX8010 ----
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index b357f1a5..5737332 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -20,6 +20,12 @@
  *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  */
 
+#ifdef __KERNEL__
+#include <linux/types.h>
+#else
+#include <stdint.h>
+#endif
+
 /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
 #define HDSPM_MAX_CHANNELS      64
 
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 0345e53..e9b8746 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -49,8 +49,6 @@
 	struct snd_pcm *pcm;
 
 	list_for_each_entry(pcm, &snd_pcm_devices, list) {
-		if (pcm->internal)
-			continue;
 		if (pcm->card == card && pcm->device == device)
 			return pcm;
 	}
@@ -62,8 +60,6 @@
 	struct snd_pcm *pcm;
 
 	list_for_each_entry(pcm, &snd_pcm_devices, list) {
-		if (pcm->internal)
-			continue;
 		if (pcm->card == card && pcm->device > device)
 			return pcm->device;
 		else if (pcm->card->number > card->number)
@@ -76,6 +72,9 @@
 {
 	struct snd_pcm *pcm;
 
+	if (newpcm->internal)
+		return 0;
+
 	list_for_each_entry(pcm, &snd_pcm_devices, list) {
 		if (pcm->card == newpcm->card && pcm->device == newpcm->device)
 			return -EBUSY;
@@ -782,6 +781,9 @@
 	pcm->card = card;
 	pcm->device = device;
 	pcm->internal = internal;
+	mutex_init(&pcm->open_mutex);
+	init_waitqueue_head(&pcm->open_wait);
+	INIT_LIST_HEAD(&pcm->list);
 	if (id)
 		strlcpy(pcm->id, id, sizeof(pcm->id));
 	if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) {
@@ -792,8 +794,6 @@
 		snd_pcm_free(pcm);
 		return err;
 	}
-	mutex_init(&pcm->open_mutex);
-	init_waitqueue_head(&pcm->open_wait);
 	if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) {
 		snd_pcm_free(pcm);
 		return err;
@@ -888,8 +888,9 @@
 
 	if (!pcm)
 		return 0;
-	list_for_each_entry(notify, &snd_pcm_notify_list, list) {
-		notify->n_unregister(pcm);
+	if (!pcm->internal) {
+		list_for_each_entry(notify, &snd_pcm_notify_list, list)
+			notify->n_unregister(pcm);
 	}
 	if (pcm->private_free)
 		pcm->private_free(pcm);
@@ -919,6 +920,9 @@
 
 	if (snd_BUG_ON(!pcm || !rsubstream))
 		return -ENXIO;
+	if (snd_BUG_ON(stream != SNDRV_PCM_STREAM_PLAYBACK &&
+		       stream != SNDRV_PCM_STREAM_CAPTURE))
+		return -EINVAL;
 	*rsubstream = NULL;
 	pstr = &pcm->streams[stream];
 	if (pstr->substream == NULL || pstr->substream_count == 0)
@@ -927,25 +931,14 @@
 	card = pcm->card;
 	prefer_subdevice = snd_ctl_get_preferred_subdevice(card, SND_CTL_SUBDEV_PCM);
 
-	switch (stream) {
-	case SNDRV_PCM_STREAM_PLAYBACK:
-		if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) {
-			for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next) {
-				if (SUBSTREAM_BUSY(substream))
-					return -EAGAIN;
-			}
+	if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) {
+		int opposite = !stream;
+
+		for (substream = pcm->streams[opposite].substream; substream;
+		     substream = substream->next) {
+			if (SUBSTREAM_BUSY(substream))
+				return -EAGAIN;
 		}
-		break;
-	case SNDRV_PCM_STREAM_CAPTURE:
-		if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) {
-			for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) {
-				if (SUBSTREAM_BUSY(substream))
-					return -EAGAIN;
-			}
-		}
-		break;
-	default:
-		return -EINVAL;
 	}
 
 	if (file->f_flags & O_APPEND) {
@@ -968,15 +961,12 @@
 		return 0;
 	}
 
-	if (prefer_subdevice >= 0) {
-		for (substream = pstr->substream; substream; substream = substream->next)
-			if (!SUBSTREAM_BUSY(substream) && substream->number == prefer_subdevice)
-				goto __ok;
-	}
-	for (substream = pstr->substream; substream; substream = substream->next)
-		if (!SUBSTREAM_BUSY(substream))
+	for (substream = pstr->substream; substream; substream = substream->next) {
+		if (!SUBSTREAM_BUSY(substream) &&
+		    (prefer_subdevice == -1 ||
+		     substream->number == prefer_subdevice))
 			break;
-      __ok:
+	}
 	if (substream == NULL)
 		return -EAGAIN;
 
@@ -1086,15 +1076,16 @@
 	if (snd_BUG_ON(!device || !device->device_data))
 		return -ENXIO;
 	pcm = device->device_data;
+	if (pcm->internal)
+		return 0;
+
 	mutex_lock(&register_mutex);
 	err = snd_pcm_add(pcm);
-	if (err) {
-		mutex_unlock(&register_mutex);
-		return err;
-	}
+	if (err)
+		goto unlock;
 	for (cidx = 0; cidx < 2; cidx++) {
 		int devtype = -1;
-		if (pcm->streams[cidx].substream == NULL || pcm->internal)
+		if (pcm->streams[cidx].substream == NULL)
 			continue;
 		switch (cidx) {
 		case SNDRV_PCM_STREAM_PLAYBACK:
@@ -1109,9 +1100,8 @@
 					  &snd_pcm_f_ops[cidx], pcm,
 					  &pcm->streams[cidx].dev);
 		if (err < 0) {
-			list_del(&pcm->list);
-			mutex_unlock(&register_mutex);
-			return err;
+			list_del_init(&pcm->list);
+			goto unlock;
 		}
 
 		for (substream = pcm->streams[cidx].substream; substream; substream = substream->next)
@@ -1121,8 +1111,9 @@
 	list_for_each_entry(notify, &snd_pcm_notify_list, list)
 		notify->n_register(pcm);
 
+ unlock:
 	mutex_unlock(&register_mutex);
-	return 0;
+	return err;
 }
 
 static int snd_pcm_dev_disconnect(struct snd_device *device)
@@ -1133,13 +1124,10 @@
 	int cidx;
 
 	mutex_lock(&register_mutex);
-	if (list_empty(&pcm->list))
-		goto unlock;
-
 	mutex_lock(&pcm->open_mutex);
 	wake_up(&pcm->open_wait);
 	list_del_init(&pcm->list);
-	for (cidx = 0; cidx < 2; cidx++)
+	for (cidx = 0; cidx < 2; cidx++) {
 		for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) {
 			snd_pcm_stream_lock_irq(substream);
 			if (substream->runtime) {
@@ -1149,18 +1137,20 @@
 			}
 			snd_pcm_stream_unlock_irq(substream);
 		}
-	list_for_each_entry(notify, &snd_pcm_notify_list, list) {
-		notify->n_disconnect(pcm);
+	}
+	if (!pcm->internal) {
+		list_for_each_entry(notify, &snd_pcm_notify_list, list)
+			notify->n_disconnect(pcm);
 	}
 	for (cidx = 0; cidx < 2; cidx++) {
-		snd_unregister_device(&pcm->streams[cidx].dev);
+		if (!pcm->internal)
+			snd_unregister_device(&pcm->streams[cidx].dev);
 		if (pcm->streams[cidx].chmap_kctl) {
 			snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl);
 			pcm->streams[cidx].chmap_kctl = NULL;
 		}
 	}
 	mutex_unlock(&pcm->open_mutex);
- unlock:
 	mutex_unlock(&register_mutex);
 	return 0;
 }
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index 2d957ba..b48b434 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -194,18 +194,30 @@
 	u32 avail_max;
 	u32 overrange;
 	s32 suspended_state;
-	u32 reserved_alignment;
+	u32 audio_tstamp_data;
 	struct compat_timespec audio_tstamp;
-	unsigned char reserved[56-sizeof(struct compat_timespec)];
+	struct compat_timespec driver_tstamp;
+	u32 audio_tstamp_accuracy;
+	unsigned char reserved[52-2*sizeof(struct compat_timespec)];
 } __attribute__((packed));
 
 
 static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream,
-				      struct snd_pcm_status32 __user *src)
+				      struct snd_pcm_status32 __user *src,
+				      bool ext)
 {
 	struct snd_pcm_status status;
 	int err;
 
+	memset(&status, 0, sizeof(status));
+	/*
+	 * with extension, parameters are read/write,
+	 * get audio_tstamp_data from user,
+	 * ignore rest of status structure
+	 */
+	if (ext && get_user(status.audio_tstamp_data,
+				(u32 __user *)(&src->audio_tstamp_data)))
+		return -EFAULT;
 	err = snd_pcm_status(substream, &status);
 	if (err < 0)
 		return err;
@@ -222,7 +234,10 @@
 	    put_user(status.avail_max, &src->avail_max) ||
 	    put_user(status.overrange, &src->overrange) ||
 	    put_user(status.suspended_state, &src->suspended_state) ||
-	    compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp))
+	    put_user(status.audio_tstamp_data, &src->audio_tstamp_data) ||
+	    compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp) ||
+	    compat_put_timespec(&status.driver_tstamp, &src->driver_tstamp) ||
+	    put_user(status.audio_tstamp_accuracy, &src->audio_tstamp_accuracy))
 		return -EFAULT;
 
 	return err;
@@ -457,6 +472,7 @@
 	SNDRV_PCM_IOCTL_HW_PARAMS32 = _IOWR('A', 0x11, struct snd_pcm_hw_params32),
 	SNDRV_PCM_IOCTL_SW_PARAMS32 = _IOWR('A', 0x13, struct snd_pcm_sw_params32),
 	SNDRV_PCM_IOCTL_STATUS32 = _IOR('A', 0x20, struct snd_pcm_status32),
+	SNDRV_PCM_IOCTL_STATUS_EXT32 = _IOWR('A', 0x24, struct snd_pcm_status32),
 	SNDRV_PCM_IOCTL_DELAY32 = _IOR('A', 0x21, s32),
 	SNDRV_PCM_IOCTL_CHANNEL_INFO32 = _IOR('A', 0x32, struct snd_pcm_channel_info32),
 	SNDRV_PCM_IOCTL_REWIND32 = _IOW('A', 0x46, u32),
@@ -517,7 +533,9 @@
 	case SNDRV_PCM_IOCTL_SW_PARAMS32:
 		return snd_pcm_ioctl_sw_params_compat(substream, argp);
 	case SNDRV_PCM_IOCTL_STATUS32:
-		return snd_pcm_status_user_compat(substream, argp);
+		return snd_pcm_status_user_compat(substream, argp, false);
+	case SNDRV_PCM_IOCTL_STATUS_EXT32:
+		return snd_pcm_status_user_compat(substream, argp, true);
 	case SNDRV_PCM_IOCTL_SYNC_PTR32:
 		return snd_pcm_ioctl_sync_ptr_compat(substream, argp);
 	case SNDRV_PCM_IOCTL_CHANNEL_INFO32:
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 6542c40..fba365a 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -289,7 +289,7 @@
  *
  * The function should usually be called from the pcm open callback. Note that
  * this function will use private_data field of the substream's runtime. So it
- * is not availabe to your pcm driver implementation.
+ * is not available to your pcm driver implementation.
  */
 int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
 	struct dma_chan *chan)
@@ -328,7 +328,7 @@
  * This function will request a DMA channel using the passed filter function and
  * data. The function should usually be called from the pcm open callback. Note
  * that this function will use private_data field of the substream's runtime. So
- * it is not availabe to your pcm driver implementation.
+ * it is not available to your pcm driver implementation.
  */
 int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
 	dma_filter_fn filter_fn, void *filter_data)
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index ffd6560..ac6b33f 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -232,6 +232,49 @@
 	return 0;
 }
 
+static void update_audio_tstamp(struct snd_pcm_substream *substream,
+				struct timespec *curr_tstamp,
+				struct timespec *audio_tstamp)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	u64 audio_frames, audio_nsecs;
+	struct timespec driver_tstamp;
+
+	if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE)
+		return;
+
+	if (!(substream->ops->get_time_info) ||
+		(runtime->audio_tstamp_report.actual_type ==
+			SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) {
+
+		/*
+		 * provide audio timestamp derived from pointer position
+		 * add delay only if requested
+		 */
+
+		audio_frames = runtime->hw_ptr_wrap + runtime->status->hw_ptr;
+
+		if (runtime->audio_tstamp_config.report_delay) {
+			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+				audio_frames -=  runtime->delay;
+			else
+				audio_frames +=  runtime->delay;
+		}
+		audio_nsecs = div_u64(audio_frames * 1000000000LL,
+				runtime->rate);
+		*audio_tstamp = ns_to_timespec(audio_nsecs);
+	}
+	runtime->status->audio_tstamp = *audio_tstamp;
+	runtime->status->tstamp = *curr_tstamp;
+
+	/*
+	 * re-take a driver timestamp to let apps detect if the reference tstamp
+	 * read by low-level hardware was provided with a delay
+	 */
+	snd_pcm_gettime(substream->runtime, (struct timespec *)&driver_tstamp);
+	runtime->driver_tstamp = driver_tstamp;
+}
+
 static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 				  unsigned int in_interrupt)
 {
@@ -256,11 +299,18 @@
 	pos = substream->ops->pointer(substream);
 	curr_jiffies = jiffies;
 	if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) {
-		snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp);
+		if ((substream->ops->get_time_info) &&
+			(runtime->audio_tstamp_config.type_requested != SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) {
+			substream->ops->get_time_info(substream, &curr_tstamp,
+						&audio_tstamp,
+						&runtime->audio_tstamp_config,
+						&runtime->audio_tstamp_report);
 
-		if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) &&
-			(substream->ops->wall_clock))
-			substream->ops->wall_clock(substream, &audio_tstamp);
+			/* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */
+			if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)
+				snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp);
+		} else
+			snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp);
 	}
 
 	if (pos == SNDRV_PCM_POS_XRUN) {
@@ -403,8 +453,10 @@
 	}
 
  no_delta_check:
-	if (runtime->status->hw_ptr == new_hw_ptr)
+	if (runtime->status->hw_ptr == new_hw_ptr) {
+		update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp);
 		return 0;
+	}
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
 	    runtime->silence_size > 0)
@@ -426,30 +478,8 @@
 		snd_BUG_ON(crossed_boundary != 1);
 		runtime->hw_ptr_wrap += runtime->boundary;
 	}
-	if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) {
-		runtime->status->tstamp = curr_tstamp;
 
-		if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) {
-			/*
-			 * no wall clock available, provide audio timestamp
-			 * derived from pointer position+delay
-			 */
-			u64 audio_frames, audio_nsecs;
-
-			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-				audio_frames = runtime->hw_ptr_wrap
-					+ runtime->status->hw_ptr
-					- runtime->delay;
-			else
-				audio_frames = runtime->hw_ptr_wrap
-					+ runtime->status->hw_ptr
-					+ runtime->delay;
-			audio_nsecs = div_u64(audio_frames * 1000000000LL,
-					runtime->rate);
-			audio_tstamp = ns_to_timespec(audio_nsecs);
-		}
-		runtime->status->audio_tstamp = audio_tstamp;
-	}
+	update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp);
 
 	return snd_pcm_update_state(substream, runtime);
 }
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 279e24f..abe1e81 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -707,6 +707,23 @@
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
 	snd_pcm_stream_lock_irq(substream);
+
+	snd_pcm_unpack_audio_tstamp_config(status->audio_tstamp_data,
+					&runtime->audio_tstamp_config);
+
+	/* backwards compatible behavior */
+	if (runtime->audio_tstamp_config.type_requested ==
+		SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT) {
+		if (runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)
+			runtime->audio_tstamp_config.type_requested =
+				SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK;
+		else
+			runtime->audio_tstamp_config.type_requested =
+				SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT;
+		runtime->audio_tstamp_report.valid = 0;
+	} else
+		runtime->audio_tstamp_report.valid = 1;
+
 	status->state = runtime->status->state;
 	status->suspended_state = runtime->status->suspended_state;
 	if (status->state == SNDRV_PCM_STATE_OPEN)
@@ -716,8 +733,15 @@
 		snd_pcm_update_hw_ptr(substream);
 		if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) {
 			status->tstamp = runtime->status->tstamp;
+			status->driver_tstamp = runtime->driver_tstamp;
 			status->audio_tstamp =
 				runtime->status->audio_tstamp;
+			if (runtime->audio_tstamp_report.valid == 1)
+				/* backwards compatibility, no report provided in COMPAT mode */
+				snd_pcm_pack_audio_tstamp_report(&status->audio_tstamp_data,
+								&status->audio_tstamp_accuracy,
+								&runtime->audio_tstamp_report);
+
 			goto _tstamp_end;
 		}
 	} else {
@@ -753,12 +777,21 @@
 }
 
 static int snd_pcm_status_user(struct snd_pcm_substream *substream,
-			       struct snd_pcm_status __user * _status)
+			       struct snd_pcm_status __user * _status,
+			       bool ext)
 {
 	struct snd_pcm_status status;
 	int res;
-	
+
 	memset(&status, 0, sizeof(status));
+	/*
+	 * with extension, parameters are read/write,
+	 * get audio_tstamp_data from user,
+	 * ignore rest of status structure
+	 */
+	if (ext && get_user(status.audio_tstamp_data,
+				(u32 __user *)(&_status->audio_tstamp_data)))
+		return -EFAULT;
 	res = snd_pcm_status(substream, &status);
 	if (res < 0)
 		return res;
@@ -2725,7 +2758,9 @@
 	case SNDRV_PCM_IOCTL_SW_PARAMS:
 		return snd_pcm_sw_params_user(substream, arg);
 	case SNDRV_PCM_IOCTL_STATUS:
-		return snd_pcm_status_user(substream, arg);
+		return snd_pcm_status_user(substream, arg, false);
+	case SNDRV_PCM_IOCTL_STATUS_EXT:
+		return snd_pcm_status_user(substream, arg, true);
 	case SNDRV_PCM_IOCTL_CHANNEL_INFO:
 		return snd_pcm_channel_info_user(substream, arg);
 	case SNDRV_PCM_IOCTL_PREPARE:
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 16d4267..72873a4 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -65,15 +65,20 @@
  * module interface
  */
 
+static struct snd_seq_driver seq_oss_synth_driver = {
+	.driver = {
+		.name = KBUILD_MODNAME,
+		.probe = snd_seq_oss_synth_probe,
+		.remove = snd_seq_oss_synth_remove,
+	},
+	.id = SNDRV_SEQ_DEV_ID_OSS,
+	.argsize = sizeof(struct snd_seq_oss_reg),
+};
+
 static int __init alsa_seq_oss_init(void)
 {
 	int rc;
-	static struct snd_seq_dev_ops ops = {
-		snd_seq_oss_synth_register,
-		snd_seq_oss_synth_unregister,
-	};
 
-	snd_seq_autoload_lock();
 	if ((rc = register_device()) < 0)
 		goto error;
 	if ((rc = register_proc()) < 0) {
@@ -86,8 +91,8 @@
 		goto error;
 	}
 
-	if ((rc = snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OSS, &ops,
-						 sizeof(struct snd_seq_oss_reg))) < 0) {
+	rc = snd_seq_driver_register(&seq_oss_synth_driver);
+	if (rc < 0) {
 		snd_seq_oss_delete_client();
 		unregister_proc();
 		unregister_device();
@@ -98,13 +103,12 @@
 	snd_seq_oss_synth_init();
 
  error:
-	snd_seq_autoload_unlock();
 	return rc;
 }
 
 static void __exit alsa_seq_oss_exit(void)
 {
-	snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OSS);
+	snd_seq_driver_unregister(&seq_oss_synth_driver);
 	snd_seq_oss_delete_client();
 	unregister_proc();
 	unregister_device();
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 701feb7..835edc8 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -98,8 +98,9 @@
  * registration of the synth device
  */
 int
-snd_seq_oss_synth_register(struct snd_seq_device *dev)
+snd_seq_oss_synth_probe(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	int i;
 	struct seq_oss_synth *rec;
 	struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev);
@@ -149,8 +150,9 @@
 
 
 int
-snd_seq_oss_synth_unregister(struct snd_seq_device *dev)
+snd_seq_oss_synth_remove(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	int index;
 	struct seq_oss_synth *rec = dev->driver_data;
 	unsigned long flags;
diff --git a/sound/core/seq/oss/seq_oss_synth.h b/sound/core/seq/oss/seq_oss_synth.h
index dbdfcbb..74ac55f 100644
--- a/sound/core/seq/oss/seq_oss_synth.h
+++ b/sound/core/seq/oss/seq_oss_synth.h
@@ -28,8 +28,8 @@
 #include <sound/seq_device.h>
 
 void snd_seq_oss_synth_init(void);
-int snd_seq_oss_synth_register(struct snd_seq_device *dev);
-int snd_seq_oss_synth_unregister(struct snd_seq_device *dev);
+int snd_seq_oss_synth_probe(struct device *dev);
+int snd_seq_oss_synth_remove(struct device *dev);
 void snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp);
 void snd_seq_oss_synth_setup_midi(struct seq_oss_devinfo *dp);
 void snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp);
diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c
index 0631bda..355b342 100644
--- a/sound/core/seq/seq_device.c
+++ b/sound/core/seq/seq_device.c
@@ -36,6 +36,7 @@
  *
  */
 
+#include <linux/device.h>
 #include <linux/init.h>
 #include <linux/module.h>
 #include <sound/core.h>
@@ -51,140 +52,78 @@
 MODULE_DESCRIPTION("ALSA sequencer device management");
 MODULE_LICENSE("GPL");
 
-/* driver state */
-#define DRIVER_EMPTY		0
-#define DRIVER_LOADED		(1<<0)
-#define DRIVER_REQUESTED	(1<<1)
-#define DRIVER_LOCKED		(1<<2)
-#define DRIVER_REQUESTING	(1<<3)
+/*
+ * bus definition
+ */
+static int snd_seq_bus_match(struct device *dev, struct device_driver *drv)
+{
+	struct snd_seq_device *sdev = to_seq_dev(dev);
+	struct snd_seq_driver *sdrv = to_seq_drv(drv);
 
-struct ops_list {
-	char id[ID_LEN];	/* driver id */
-	int driver;		/* driver state */
-	int used;		/* reference counter */
-	int argsize;		/* argument size */
+	return strcmp(sdrv->id, sdev->id) == 0 &&
+		sdrv->argsize == sdev->argsize;
+}
 
-	/* operators */
-	struct snd_seq_dev_ops ops;
-
-	/* registered devices */
-	struct list_head dev_list;	/* list of devices */
-	int num_devices;	/* number of associated devices */
-	int num_init_devices;	/* number of initialized devices */
-	struct mutex reg_mutex;
-
-	struct list_head list;	/* next driver */
+static struct bus_type snd_seq_bus_type = {
+	.name = "snd_seq",
+	.match = snd_seq_bus_match,
 };
 
-
-static LIST_HEAD(opslist);
-static int num_ops;
-static DEFINE_MUTEX(ops_mutex);
+/*
+ * proc interface -- just for compatibility
+ */
 #ifdef CONFIG_PROC_FS
 static struct snd_info_entry *info_entry;
-#endif
 
-/*
- * prototypes
- */
-static int snd_seq_device_free(struct snd_seq_device *dev);
-static int snd_seq_device_dev_free(struct snd_device *device);
-static int snd_seq_device_dev_register(struct snd_device *device);
-static int snd_seq_device_dev_disconnect(struct snd_device *device);
+static int print_dev_info(struct device *dev, void *data)
+{
+	struct snd_seq_device *sdev = to_seq_dev(dev);
+	struct snd_info_buffer *buffer = data;
 
-static int init_device(struct snd_seq_device *dev, struct ops_list *ops);
-static int free_device(struct snd_seq_device *dev, struct ops_list *ops);
-static struct ops_list *find_driver(char *id, int create_if_empty);
-static struct ops_list *create_driver(char *id);
-static void unlock_driver(struct ops_list *ops);
-static void remove_drivers(void);
+	snd_iprintf(buffer, "snd-%s,%s,%d\n", sdev->id,
+		    dev->driver ? "loaded" : "empty",
+		    dev->driver ? 1 : 0);
+	return 0;
+}
 
-/*
- * show all drivers and their status
- */
-
-#ifdef CONFIG_PROC_FS
 static void snd_seq_device_info(struct snd_info_entry *entry,
 				struct snd_info_buffer *buffer)
 {
-	struct ops_list *ops;
-
-	mutex_lock(&ops_mutex);
-	list_for_each_entry(ops, &opslist, list) {
-		snd_iprintf(buffer, "snd-%s%s%s%s,%d\n",
-				ops->id,
-				ops->driver & DRIVER_LOADED ? ",loaded" : (ops->driver == DRIVER_EMPTY ? ",empty" : ""),
-				ops->driver & DRIVER_REQUESTED ? ",requested" : "",
-				ops->driver & DRIVER_LOCKED ? ",locked" : "",
-				ops->num_devices);
-	}
-	mutex_unlock(&ops_mutex);
+	bus_for_each_dev(&snd_seq_bus_type, NULL, buffer, print_dev_info);
 }
 #endif
- 
+
 /*
  * load all registered drivers (called from seq_clientmgr.c)
  */
 
 #ifdef CONFIG_MODULES
-/* avoid auto-loading during module_init() */
+/* flag to block auto-loading */
 static atomic_t snd_seq_in_init = ATOMIC_INIT(1); /* blocked as default */
-void snd_seq_autoload_lock(void)
+
+static int request_seq_drv(struct device *dev, void *data)
 {
-	atomic_inc(&snd_seq_in_init);
+	struct snd_seq_device *sdev = to_seq_dev(dev);
+
+	if (!dev->driver)
+		request_module("snd-%s", sdev->id);
+	return 0;
 }
 
-void snd_seq_autoload_unlock(void)
-{
-	atomic_dec(&snd_seq_in_init);
-}
-
-static void autoload_drivers(void)
+static void autoload_drivers(struct work_struct *work)
 {
 	/* avoid reentrance */
-	if (atomic_inc_return(&snd_seq_in_init) == 1) {
-		struct ops_list *ops;
-
-		mutex_lock(&ops_mutex);
-		list_for_each_entry(ops, &opslist, list) {
-			if ((ops->driver & DRIVER_REQUESTING) &&
-			    !(ops->driver & DRIVER_REQUESTED)) {
-				ops->used++;
-				mutex_unlock(&ops_mutex);
-				ops->driver |= DRIVER_REQUESTED;
-				request_module("snd-%s", ops->id);
-				mutex_lock(&ops_mutex);
-				ops->used--;
-			}
-		}
-		mutex_unlock(&ops_mutex);
-	}
+	if (atomic_inc_return(&snd_seq_in_init) == 1)
+		bus_for_each_dev(&snd_seq_bus_type, NULL, NULL,
+				 request_seq_drv);
 	atomic_dec(&snd_seq_in_init);
 }
 
-static void call_autoload(struct work_struct *work)
-{
-	autoload_drivers();
-}
-
-static DECLARE_WORK(autoload_work, call_autoload);
-
-static void try_autoload(struct ops_list *ops)
-{
-	if (!ops->driver) {
-		ops->driver |= DRIVER_REQUESTING;
-		schedule_work(&autoload_work);
-	}
-}
+static DECLARE_WORK(autoload_work, autoload_drivers);
 
 static void queue_autoload_drivers(void)
 {
-	struct ops_list *ops;
-
-	mutex_lock(&ops_mutex);
-	list_for_each_entry(ops, &opslist, list)
-		try_autoload(ops);
-	mutex_unlock(&ops_mutex);
+	schedule_work(&autoload_work);
 }
 
 void snd_seq_autoload_init(void)
@@ -195,16 +134,63 @@
 	queue_autoload_drivers();
 #endif
 }
-#else
-#define try_autoload(ops) /* NOP */
-#endif
+EXPORT_SYMBOL(snd_seq_autoload_init);
+
+void snd_seq_autoload_exit(void)
+{
+	atomic_inc(&snd_seq_in_init);
+}
+EXPORT_SYMBOL(snd_seq_autoload_exit);
 
 void snd_seq_device_load_drivers(void)
 {
-#ifdef CONFIG_MODULES
 	queue_autoload_drivers();
 	flush_work(&autoload_work);
+}
+EXPORT_SYMBOL(snd_seq_device_load_drivers);
+#else
+#define queue_autoload_drivers() /* NOP */
 #endif
+
+/*
+ * device management
+ */
+static int snd_seq_device_dev_free(struct snd_device *device)
+{
+	struct snd_seq_device *dev = device->device_data;
+
+	put_device(&dev->dev);
+	return 0;
+}
+
+static int snd_seq_device_dev_register(struct snd_device *device)
+{
+	struct snd_seq_device *dev = device->device_data;
+	int err;
+
+	err = device_add(&dev->dev);
+	if (err < 0)
+		return err;
+	if (!dev->dev.driver)
+		queue_autoload_drivers();
+	return 0;
+}
+
+static int snd_seq_device_dev_disconnect(struct snd_device *device)
+{
+	struct snd_seq_device *dev = device->device_data;
+
+	device_del(&dev->dev);
+	return 0;
+}
+
+static void snd_seq_dev_release(struct device *dev)
+{
+	struct snd_seq_device *sdev = to_seq_dev(dev);
+
+	if (sdev->private_free)
+		sdev->private_free(sdev);
+	kfree(sdev);
 }
 
 /*
@@ -214,11 +200,10 @@
  * id = id of driver
  * result = return pointer (NULL allowed if unnecessary)
  */
-int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize,
-		       struct snd_seq_device **result)
+int snd_seq_device_new(struct snd_card *card, int device, const char *id,
+		       int argsize, struct snd_seq_device **result)
 {
 	struct snd_seq_device *dev;
-	struct ops_list *ops;
 	int err;
 	static struct snd_device_ops dops = {
 		.dev_free = snd_seq_device_dev_free,
@@ -232,347 +217,60 @@
 	if (snd_BUG_ON(!id))
 		return -EINVAL;
 
-	ops = find_driver(id, 1);
-	if (ops == NULL)
+	dev = kzalloc(sizeof(*dev) + argsize, GFP_KERNEL);
+	if (!dev)
 		return -ENOMEM;
 
-	dev = kzalloc(sizeof(*dev)*2 + argsize, GFP_KERNEL);
-	if (dev == NULL) {
-		unlock_driver(ops);
-		return -ENOMEM;
-	}
-
 	/* set up device info */
 	dev->card = card;
 	dev->device = device;
-	strlcpy(dev->id, id, sizeof(dev->id));
+	dev->id = id;
 	dev->argsize = argsize;
-	dev->status = SNDRV_SEQ_DEVICE_FREE;
+
+	device_initialize(&dev->dev);
+	dev->dev.parent = &card->card_dev;
+	dev->dev.bus = &snd_seq_bus_type;
+	dev->dev.release = snd_seq_dev_release;
+	dev_set_name(&dev->dev, "%s-%d-%d", dev->id, card->number, device);
 
 	/* add this device to the list */
-	mutex_lock(&ops->reg_mutex);
-	list_add_tail(&dev->list, &ops->dev_list);
-	ops->num_devices++;
-	mutex_unlock(&ops->reg_mutex);
-
-	if ((err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops)) < 0) {
-		snd_seq_device_free(dev);
+	err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops);
+	if (err < 0) {
+		put_device(&dev->dev);
 		return err;
 	}
 	
-	try_autoload(ops);
-	unlock_driver(ops);
-
 	if (result)
 		*result = dev;
 
 	return 0;
 }
+EXPORT_SYMBOL(snd_seq_device_new);
 
 /*
- * free the existing device
+ * driver registration
  */
-static int snd_seq_device_free(struct snd_seq_device *dev)
+int __snd_seq_driver_register(struct snd_seq_driver *drv, struct module *mod)
 {
-	struct ops_list *ops;
-
-	if (snd_BUG_ON(!dev))
+	if (WARN_ON(!drv->driver.name || !drv->id))
 		return -EINVAL;
-
-	ops = find_driver(dev->id, 0);
-	if (ops == NULL)
-		return -ENXIO;
-
-	/* remove the device from the list */
-	mutex_lock(&ops->reg_mutex);
-	list_del(&dev->list);
-	ops->num_devices--;
-	mutex_unlock(&ops->reg_mutex);
-
-	free_device(dev, ops);
-	if (dev->private_free)
-		dev->private_free(dev);
-	kfree(dev);
-
-	unlock_driver(ops);
-
-	return 0;
+	drv->driver.bus = &snd_seq_bus_type;
+	drv->driver.owner = mod;
+	return driver_register(&drv->driver);
 }
+EXPORT_SYMBOL_GPL(__snd_seq_driver_register);
 
-static int snd_seq_device_dev_free(struct snd_device *device)
+void snd_seq_driver_unregister(struct snd_seq_driver *drv)
 {
-	struct snd_seq_device *dev = device->device_data;
-	return snd_seq_device_free(dev);
+	driver_unregister(&drv->driver);
 }
-
-/*
- * register the device
- */
-static int snd_seq_device_dev_register(struct snd_device *device)
-{
-	struct snd_seq_device *dev = device->device_data;
-	struct ops_list *ops;
-
-	ops = find_driver(dev->id, 0);
-	if (ops == NULL)
-		return -ENOENT;
-
-	/* initialize this device if the corresponding driver was
-	 * already loaded
-	 */
-	if (ops->driver & DRIVER_LOADED)
-		init_device(dev, ops);
-
-	unlock_driver(ops);
-	return 0;
-}
-
-/*
- * disconnect the device
- */
-static int snd_seq_device_dev_disconnect(struct snd_device *device)
-{
-	struct snd_seq_device *dev = device->device_data;
-	struct ops_list *ops;
-
-	ops = find_driver(dev->id, 0);
-	if (ops == NULL)
-		return -ENOENT;
-
-	free_device(dev, ops);
-
-	unlock_driver(ops);
-	return 0;
-}
-
-/*
- * register device driver
- * id = driver id
- * entry = driver operators - duplicated to each instance
- */
-int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry,
-				   int argsize)
-{
-	struct ops_list *ops;
-	struct snd_seq_device *dev;
-
-	if (id == NULL || entry == NULL ||
-	    entry->init_device == NULL || entry->free_device == NULL)
-		return -EINVAL;
-
-	ops = find_driver(id, 1);
-	if (ops == NULL)
-		return -ENOMEM;
-	if (ops->driver & DRIVER_LOADED) {
-		pr_warn("ALSA: seq: driver_register: driver '%s' already exists\n", id);
-		unlock_driver(ops);
-		return -EBUSY;
-	}
-
-	mutex_lock(&ops->reg_mutex);
-	/* copy driver operators */
-	ops->ops = *entry;
-	ops->driver |= DRIVER_LOADED;
-	ops->argsize = argsize;
-
-	/* initialize existing devices if necessary */
-	list_for_each_entry(dev, &ops->dev_list, list) {
-		init_device(dev, ops);
-	}
-	mutex_unlock(&ops->reg_mutex);
-
-	unlock_driver(ops);
-
-	return 0;
-}
-
-
-/*
- * create driver record
- */
-static struct ops_list * create_driver(char *id)
-{
-	struct ops_list *ops;
-
-	ops = kzalloc(sizeof(*ops), GFP_KERNEL);
-	if (ops == NULL)
-		return ops;
-
-	/* set up driver entry */
-	strlcpy(ops->id, id, sizeof(ops->id));
-	mutex_init(&ops->reg_mutex);
-	/*
-	 * The ->reg_mutex locking rules are per-driver, so we create
-	 * separate per-driver lock classes:
-	 */
-	lockdep_set_class(&ops->reg_mutex, (struct lock_class_key *)id);
-
-	ops->driver = DRIVER_EMPTY;
-	INIT_LIST_HEAD(&ops->dev_list);
-	/* lock this instance */
-	ops->used = 1;
-
-	/* register driver entry */
-	mutex_lock(&ops_mutex);
-	list_add_tail(&ops->list, &opslist);
-	num_ops++;
-	mutex_unlock(&ops_mutex);
-
-	return ops;
-}
-
-
-/*
- * unregister the specified driver
- */
-int snd_seq_device_unregister_driver(char *id)
-{
-	struct ops_list *ops;
-	struct snd_seq_device *dev;
-
-	ops = find_driver(id, 0);
-	if (ops == NULL)
-		return -ENXIO;
-	if (! (ops->driver & DRIVER_LOADED) ||
-	    (ops->driver & DRIVER_LOCKED)) {
-		pr_err("ALSA: seq: driver_unregister: cannot unload driver '%s': status=%x\n",
-			   id, ops->driver);
-		unlock_driver(ops);
-		return -EBUSY;
-	}
-
-	/* close and release all devices associated with this driver */
-	mutex_lock(&ops->reg_mutex);
-	ops->driver |= DRIVER_LOCKED; /* do not remove this driver recursively */
-	list_for_each_entry(dev, &ops->dev_list, list) {
-		free_device(dev, ops);
-	}
-
-	ops->driver = 0;
-	if (ops->num_init_devices > 0)
-		pr_err("ALSA: seq: free_driver: init_devices > 0!! (%d)\n",
-			   ops->num_init_devices);
-	mutex_unlock(&ops->reg_mutex);
-
-	unlock_driver(ops);
-
-	/* remove empty driver entries */
-	remove_drivers();
-
-	return 0;
-}
-
-
-/*
- * remove empty driver entries
- */
-static void remove_drivers(void)
-{
-	struct list_head *head;
-
-	mutex_lock(&ops_mutex);
-	head = opslist.next;
-	while (head != &opslist) {
-		struct ops_list *ops = list_entry(head, struct ops_list, list);
-		if (! (ops->driver & DRIVER_LOADED) &&
-		    ops->used == 0 && ops->num_devices == 0) {
-			head = head->next;
-			list_del(&ops->list);
-			kfree(ops);
-			num_ops--;
-		} else
-			head = head->next;
-	}
-	mutex_unlock(&ops_mutex);
-}
-
-/*
- * initialize the device - call init_device operator
- */
-static int init_device(struct snd_seq_device *dev, struct ops_list *ops)
-{
-	if (! (ops->driver & DRIVER_LOADED))
-		return 0; /* driver is not loaded yet */
-	if (dev->status != SNDRV_SEQ_DEVICE_FREE)
-		return 0; /* already initialized */
-	if (ops->argsize != dev->argsize) {
-		pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n",
-			   dev->name, ops->id, ops->argsize, dev->argsize);
-		return -EINVAL;
-	}
-	if (ops->ops.init_device(dev) >= 0) {
-		dev->status = SNDRV_SEQ_DEVICE_REGISTERED;
-		ops->num_init_devices++;
-	} else {
-		pr_err("ALSA: seq: init_device failed: %s: %s\n",
-			   dev->name, dev->id);
-	}
-
-	return 0;
-}
-
-/*
- * release the device - call free_device operator
- */
-static int free_device(struct snd_seq_device *dev, struct ops_list *ops)
-{
-	int result;
-
-	if (! (ops->driver & DRIVER_LOADED))
-		return 0; /* driver is not loaded yet */
-	if (dev->status != SNDRV_SEQ_DEVICE_REGISTERED)
-		return 0; /* not registered */
-	if (ops->argsize != dev->argsize) {
-		pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n",
-			   dev->name, ops->id, ops->argsize, dev->argsize);
-		return -EINVAL;
-	}
-	if ((result = ops->ops.free_device(dev)) >= 0 || result == -ENXIO) {
-		dev->status = SNDRV_SEQ_DEVICE_FREE;
-		dev->driver_data = NULL;
-		ops->num_init_devices--;
-	} else {
-		pr_err("ALSA: seq: free_device failed: %s: %s\n",
-			   dev->name, dev->id);
-	}
-
-	return 0;
-}
-
-/*
- * find the matching driver with given id
- */
-static struct ops_list * find_driver(char *id, int create_if_empty)
-{
-	struct ops_list *ops;
-
-	mutex_lock(&ops_mutex);
-	list_for_each_entry(ops, &opslist, list) {
-		if (strcmp(ops->id, id) == 0) {
-			ops->used++;
-			mutex_unlock(&ops_mutex);
-			return ops;
-		}
-	}
-	mutex_unlock(&ops_mutex);
-	if (create_if_empty)
-		return create_driver(id);
-	return NULL;
-}
-
-static void unlock_driver(struct ops_list *ops)
-{
-	mutex_lock(&ops_mutex);
-	ops->used--;
-	mutex_unlock(&ops_mutex);
-}
-
+EXPORT_SYMBOL_GPL(snd_seq_driver_unregister);
 
 /*
  * module part
  */
 
-static int __init alsa_seq_device_init(void)
+static int __init seq_dev_proc_init(void)
 {
 #ifdef CONFIG_PROC_FS
 	info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers",
@@ -589,28 +287,29 @@
 	return 0;
 }
 
+static int __init alsa_seq_device_init(void)
+{
+	int err;
+
+	err = bus_register(&snd_seq_bus_type);
+	if (err < 0)
+		return err;
+	err = seq_dev_proc_init();
+	if (err < 0)
+		bus_unregister(&snd_seq_bus_type);
+	return err;
+}
+
 static void __exit alsa_seq_device_exit(void)
 {
 #ifdef CONFIG_MODULES
 	cancel_work_sync(&autoload_work);
 #endif
-	remove_drivers();
 #ifdef CONFIG_PROC_FS
 	snd_info_free_entry(info_entry);
 #endif
-	if (num_ops)
-		pr_err("ALSA: seq: drivers not released (%d)\n", num_ops);
+	bus_unregister(&snd_seq_bus_type);
 }
 
 module_init(alsa_seq_device_init)
 module_exit(alsa_seq_device_exit)
-
-EXPORT_SYMBOL(snd_seq_device_load_drivers);
-EXPORT_SYMBOL(snd_seq_device_new);
-EXPORT_SYMBOL(snd_seq_device_register_driver);
-EXPORT_SYMBOL(snd_seq_device_unregister_driver);
-#ifdef CONFIG_MODULES
-EXPORT_SYMBOL(snd_seq_autoload_init);
-EXPORT_SYMBOL(snd_seq_autoload_lock);
-EXPORT_SYMBOL(snd_seq_autoload_unlock);
-#endif
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index 5d905d9..d3a2ec4 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -214,11 +214,7 @@
 
 static int __init alsa_seq_dummy_init(void)
 {
-	int err;
-	snd_seq_autoload_lock();
-	err = register_client();
-	snd_seq_autoload_unlock();
-	return err;
+	return register_client();
 }
 
 static void __exit alsa_seq_dummy_exit(void)
diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c
index 68fec77..5dd0ee2 100644
--- a/sound/core/seq/seq_midi.c
+++ b/sound/core/seq/seq_midi.c
@@ -273,8 +273,9 @@
 
 /* register new midi synth port */
 static int
-snd_seq_midisynth_register_port(struct snd_seq_device *dev)
+snd_seq_midisynth_probe(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct seq_midisynth_client *client;
 	struct seq_midisynth *msynth, *ms;
 	struct snd_seq_port_info *port;
@@ -427,8 +428,9 @@
 
 /* release midi synth port */
 static int
-snd_seq_midisynth_unregister_port(struct snd_seq_device *dev)
+snd_seq_midisynth_remove(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct seq_midisynth_client *client;
 	struct seq_midisynth *msynth;
 	struct snd_card *card = dev->card;
@@ -457,24 +459,14 @@
 	return 0;
 }
 
+static struct snd_seq_driver seq_midisynth_driver = {
+	.driver = {
+		.name = KBUILD_MODNAME,
+		.probe = snd_seq_midisynth_probe,
+		.remove = snd_seq_midisynth_remove,
+	},
+	.id = SNDRV_SEQ_DEV_ID_MIDISYNTH,
+	.argsize = 0,
+};
 
-static int __init alsa_seq_midi_init(void)
-{
-	static struct snd_seq_dev_ops ops = {
-		snd_seq_midisynth_register_port,
-		snd_seq_midisynth_unregister_port,
-	};
-	memset(&synths, 0, sizeof(synths));
-	snd_seq_autoload_lock();
-	snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH, &ops, 0);
-	snd_seq_autoload_unlock();
-	return 0;
-}
-
-static void __exit alsa_seq_midi_exit(void)
-{
-	snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH);
-}
-
-module_init(alsa_seq_midi_init)
-module_exit(alsa_seq_midi_exit)
+module_snd_seq_driver(seq_midisynth_driver);
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 185cec0..5fc93d0 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -186,7 +186,7 @@
 };
 
 #ifdef CONFIG_SND_DYNAMIC_MINORS
-static int snd_find_free_minor(int type)
+static int snd_find_free_minor(int type, struct snd_card *card, int dev)
 {
 	int minor;
 
@@ -209,7 +209,7 @@
 	return -EBUSY;
 }
 #else
-static int snd_kernel_minor(int type, struct snd_card *card, int dev)
+static int snd_find_free_minor(int type, struct snd_card *card, int dev)
 {
 	int minor;
 
@@ -237,6 +237,8 @@
 	}
 	if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OS_MINORS))
 		return -EINVAL;
+	if (snd_minors[minor])
+		return -EBUSY;
 	return minor;
 }
 #endif
@@ -276,13 +278,7 @@
 	preg->private_data = private_data;
 	preg->card_ptr = card;
 	mutex_lock(&sound_mutex);
-#ifdef CONFIG_SND_DYNAMIC_MINORS
-	minor = snd_find_free_minor(type);
-#else
-	minor = snd_kernel_minor(type, card, dev);
-	if (minor >= 0 && snd_minors[minor])
-		minor = -EBUSY;
-#endif
+	minor = snd_find_free_minor(type, card, dev);
 	if (minor < 0) {
 		err = minor;
 		goto error;
diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c
index a9f618e..fdae5d7 100644
--- a/sound/drivers/opl3/opl3_seq.c
+++ b/sound/drivers/opl3/opl3_seq.c
@@ -216,8 +216,9 @@
 
 /* ------------------------------ */
 
-static int snd_opl3_seq_new_device(struct snd_seq_device *dev)
+static int snd_opl3_seq_probe(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_opl3 *opl3;
 	int client, err;
 	char name[32];
@@ -257,8 +258,9 @@
 	return 0;
 }
 
-static int snd_opl3_seq_delete_device(struct snd_seq_device *dev)
+static int snd_opl3_seq_remove(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_opl3 *opl3;
 
 	opl3 = *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(dev);
@@ -275,22 +277,14 @@
 	return 0;
 }
 
-static int __init alsa_opl3_seq_init(void)
-{
-	static struct snd_seq_dev_ops ops =
-	{
-		snd_opl3_seq_new_device,
-		snd_opl3_seq_delete_device
-	};
+static struct snd_seq_driver opl3_seq_driver = {
+	.driver = {
+		.name = KBUILD_MODNAME,
+		.probe = snd_opl3_seq_probe,
+		.remove = snd_opl3_seq_remove,
+	},
+	.id = SNDRV_SEQ_DEV_ID_OPL3,
+	.argsize = sizeof(struct snd_opl3 *),
+};
 
-	return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL3, &ops,
-					      sizeof(struct snd_opl3 *));
-}
-
-static void __exit alsa_opl3_seq_exit(void)
-{
-	snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL3);
-}
-
-module_init(alsa_opl3_seq_init)
-module_exit(alsa_opl3_seq_exit)
+module_snd_seq_driver(opl3_seq_driver);
diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c
index 9919769..03d6202 100644
--- a/sound/drivers/opl4/opl4_seq.c
+++ b/sound/drivers/opl4/opl4_seq.c
@@ -124,8 +124,9 @@
 	snd_midi_channel_free_set(opl4->chset);
 }
 
-static int snd_opl4_seq_new_device(struct snd_seq_device *dev)
+static int snd_opl4_seq_probe(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_opl4 *opl4;
 	int client;
 	struct snd_seq_port_callback pcallbacks;
@@ -180,8 +181,9 @@
 	return 0;
 }
 
-static int snd_opl4_seq_delete_device(struct snd_seq_device *dev)
+static int snd_opl4_seq_remove(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_opl4 *opl4;
 
 	opl4 = *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(dev);
@@ -195,21 +197,14 @@
 	return 0;
 }
 
-static int __init alsa_opl4_synth_init(void)
-{
-	static struct snd_seq_dev_ops ops = {
-		snd_opl4_seq_new_device,
-		snd_opl4_seq_delete_device
-	};
+static struct snd_seq_driver opl4_seq_driver = {
+	.driver = {
+		.name = KBUILD_MODNAME,
+		.probe = snd_opl4_seq_probe,
+		.remove = snd_opl4_seq_remove,
+	},
+	.id = SNDRV_SEQ_DEV_ID_OPL4,
+	.argsize = sizeof(struct snd_opl4 *),
+};
 
-	return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL4, &ops,
-					      sizeof(struct snd_opl4 *));
-}
-
-static void __exit alsa_opl4_synth_exit(void)
-{
-	snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL4);
-}
-
-module_init(alsa_opl4_synth_init)
-module_exit(alsa_opl4_synth_exit)
+module_snd_seq_driver(opl4_seq_driver);
diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c
index 72332df..4aa719c 100644
--- a/sound/isa/sb/emu8000_synth.c
+++ b/sound/isa/sb/emu8000_synth.c
@@ -34,8 +34,9 @@
 /*
  * create a new hardware dependent device for Emu8000
  */
-static int snd_emu8000_new_device(struct snd_seq_device *dev)
+static int snd_emu8000_probe(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_emu8000 *hw;
 	struct snd_emux *emu;
 
@@ -93,8 +94,9 @@
 /*
  * free all resources
  */
-static int snd_emu8000_delete_device(struct snd_seq_device *dev)
+static int snd_emu8000_remove(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_emu8000 *hw;
 
 	if (dev->driver_data == NULL)
@@ -114,21 +116,14 @@
  *  INIT part
  */
 
-static int __init alsa_emu8000_init(void)
-{
-	
-	static struct snd_seq_dev_ops ops = {
-		snd_emu8000_new_device,
-		snd_emu8000_delete_device,
-	};
-	return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU8000, &ops,
-					      sizeof(struct snd_emu8000*));
-}
+static struct snd_seq_driver emu8000_driver = {
+	.driver = {
+		.name = KBUILD_MODNAME,
+		.probe = snd_emu8000_probe,
+		.remove = snd_emu8000_remove,
+	},
+	.id = SNDRV_SEQ_DEV_ID_EMU8000,
+	.argsize = sizeof(struct snd_emu8000 *),
+};
 
-static void __exit alsa_emu8000_exit(void)
-{
-	snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU8000);
-}
-
-module_init(alsa_emu8000_init)
-module_exit(alsa_emu8000_exit)
+module_snd_seq_driver(emu8000_driver);
diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c
index 607cee4..b6d19ad 100644
--- a/sound/oss/opl3.c
+++ b/sound/oss/opl3.c
@@ -666,7 +666,7 @@
 	opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data);
 
 	data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3);
-		 devc->voc[voice].keyon_byte = data;
+	devc->voc[voice].keyon_byte = data;
 	opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data);
 	if (voice_mode == 4)
 		opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data);
@@ -717,7 +717,7 @@
 
 static void opl3_command    (int io_addr, unsigned int addr, unsigned int val)
 {
-	 int i;
+	int i;
 
 	/*
 	 * The original 2-OP synth requires a quite long delay after writing to a
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c
index b47a690..57f7d25 100644
--- a/sound/oss/sb_ess.c
+++ b/sound/oss/sb_ess.c
@@ -604,7 +604,7 @@
 	ess_chgmixer (devc, 0x78, 0x03, 0x03);   /* Go */
 
 	devc->irq_mode_16 = IMODE_OUTPUT;
-		devc->intr_active_16 = 1;
+	devc->intr_active_16 = 1;
 }
 
 static void ess_audio_output_block
@@ -1183,17 +1183,12 @@
 			chip = "ES1688";
 		}
 
-	    printk ( KERN_INFO "ESS chip %s %s%s\n"
-               , chip
-               , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20
-                 ? "detected"
-                 : "specified"
-                 )
-               , ( devc->sbmo.esstype == ESSTYPE_LIKE20
-                 ? " (kernel 2.0 compatible)"
-                 : ""
-                 )
-               );
+		printk(KERN_INFO "ESS chip %s %s%s\n", chip,
+		       (devc->sbmo.esstype == ESSTYPE_DETECT ||
+			devc->sbmo.esstype == ESSTYPE_LIKE20) ?
+				"detected" : "specified",
+			devc->sbmo.esstype == ESSTYPE_LIKE20 ?
+				" (kernel 2.0 compatible)" : "");
 
 		sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f);
 	} else {
diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c
index f139028..551ee75 100644
--- a/sound/oss/sb_midi.c
+++ b/sound/oss/sb_midi.c
@@ -179,14 +179,14 @@
 	{
 		printk(KERN_WARNING "Sound Blaster:  failed to allocate MIDI memory.\n");
 		sound_unload_mididev(dev);
-		  return;
+		return;
 	}
 	memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations,
 	       sizeof(struct midi_operations));
 
 	if (owner)
-			midi_devs[dev]->owner = owner;
-	
+		midi_devs[dev]->owner = owner;
+
 	midi_devs[dev]->devc = devc;
 
 
diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c
index 9f03983..2226dda 100644
--- a/sound/oss/sys_timer.c
+++ b/sound/oss/sys_timer.c
@@ -50,29 +50,24 @@
 static void
 poll_def_tmr(unsigned long dummy)
 {
+	if (!opened)
+		return;
+	def_tmr.expires = (1) + jiffies;
+	add_timer(&def_tmr);
 
-	if (opened)
-	  {
+	if (!tmr_running)
+		return;
 
-		  {
-			  def_tmr.expires = (1) + jiffies;
-			  add_timer(&def_tmr);
-		  }
+	spin_lock(&lock);
+	tmr_ctr++;
+	curr_ticks = ticks_offs + tmr2ticks(tmr_ctr);
 
-		  if (tmr_running)
-		    {
-				spin_lock(&lock);
-			    tmr_ctr++;
-			    curr_ticks = ticks_offs + tmr2ticks(tmr_ctr);
+	if (curr_ticks >= next_event_time) {
+		next_event_time = (unsigned long) -1;
+		sequencer_timer(0);
+	}
 
-			    if (curr_ticks >= next_event_time)
-			      {
-				      next_event_time = (unsigned long) -1;
-				      sequencer_timer(0);
-			      }
-				spin_unlock(&lock);
-		    }
-	  }
+	spin_unlock(&lock);
 }
 
 static void
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 5ee2f17..5bca1a3 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -177,6 +177,7 @@
 { 0x54524123, 0xffffffff, "TR28602",		NULL,		NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)]
 { 0x54584e03, 0xffffffff, "TLV320AIC27",	NULL,		NULL },
 { 0x54584e20, 0xffffffff, "TLC320AD9xC",	NULL,		NULL },
+{ 0x56494120, 0xfffffff0, "VIA1613",		patch_vt1613,	NULL },
 { 0x56494161, 0xffffffff, "VIA1612A",		NULL,		NULL }, // modified ICE1232 with S/PDIF
 { 0x56494170, 0xffffffff, "VIA1617A",		patch_vt1617a,	NULL }, // modified VT1616 with S/PDIF
 { 0x56494182, 0xffffffff, "VIA1618",		patch_vt1618,   NULL },
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index ceaac1c..f4234ed 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -3352,6 +3352,33 @@
 }
 
 /*
+ * VIA VT1613 codec
+ */
+static const struct snd_kcontrol_new snd_ac97_controls_vt1613[] = {
+AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0),
+};
+
+static int patch_vt1613_specific(struct snd_ac97 *ac97)
+{
+	return patch_build_controls(ac97, &snd_ac97_controls_vt1613[0],
+				    ARRAY_SIZE(snd_ac97_controls_vt1613));
+};
+
+static const struct snd_ac97_build_ops patch_vt1613_ops = {
+	.build_specific	= patch_vt1613_specific
+};
+
+static int patch_vt1613(struct snd_ac97 *ac97)
+{
+	ac97->build_ops = &patch_vt1613_ops;
+
+	ac97->flags |= AC97_HAS_NO_VIDEO;
+	ac97->caps |= AC97_BC_HEADPHONE;
+
+	return 0;
+}
+
+/*
  * VIA VT1616 codec
  */
 static const struct snd_kcontrol_new snd_ac97_controls_vt1616[] = {
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index a40a2b4..33b2a0a 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1385,8 +1385,8 @@
 					.running)
 			     &&  (!chip->codecs[peer_codecs[codec_type].other2]
 					.running));
-		 }
-		 if (call_function)
+		}
+		if (call_function)
 			snd_azf3328_ctrl_enable_codecs(chip, enable);
 
 		/* ...and adjust clock, too
@@ -2126,7 +2126,8 @@
 static int
 snd_azf3328_pcm(struct snd_azf3328 *chip)
 {
-enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */
+	/* pcm devices */
+	enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS };
 
 	struct snd_pcm *pcm;
 	int err;
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 1d0f2ca..6cf464d 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -2062,7 +2062,7 @@
 		val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask;
 		if (reg.invert)
 			val = reg.mask - val;
-		 ucontrol->value.integer.value[1] = val;
+		ucontrol->value.integer.value[1] = val;
 	}
 	spin_unlock_irq(&cm->reg_lock);
 	return 0;
diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c
index 4c41c90..5457d56 100644
--- a/sound/pci/emu10k1/emu10k1_synth.c
+++ b/sound/pci/emu10k1/emu10k1_synth.c
@@ -29,8 +29,9 @@
 /*
  * create a new hardware dependent device for Emu10k1
  */
-static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev)
+static int snd_emu10k1_synth_probe(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_emux *emux;
 	struct snd_emu10k1 *hw;
 	struct snd_emu10k1_synth_arg *arg;
@@ -79,8 +80,9 @@
 	return 0;
 }
 
-static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev)
+static int snd_emu10k1_synth_remove(struct device *_dev)
 {
+	struct snd_seq_device *dev = to_seq_dev(_dev);
 	struct snd_emux *emux;
 	struct snd_emu10k1 *hw;
 	unsigned long flags;
@@ -104,21 +106,14 @@
  *  INIT part
  */
 
-static int __init alsa_emu10k1_synth_init(void)
-{
-	
-	static struct snd_seq_dev_ops ops = {
-		snd_emu10k1_synth_new_device,
-		snd_emu10k1_synth_delete_device,
-	};
-	return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, &ops,
-					      sizeof(struct snd_emu10k1_synth_arg));
-}
+static struct snd_seq_driver emu10k1_synth_driver = {
+	.driver = {
+		.name = KBUILD_MODNAME,
+		.probe = snd_emu10k1_synth_probe,
+		.remove = snd_emu10k1_synth_remove,
+	},
+	.id = SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH,
+	.argsize = sizeof(struct snd_emu10k1_synth_arg),
+};
 
-static void __exit alsa_emu10k1_synth_exit(void)
-{
-	snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH);
-}
-
-module_init(alsa_emu10k1_synth_init)
-module_exit(alsa_emu10k1_synth_exit)
+module_snd_seq_driver(emu10k1_synth_driver);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 2fe86d2..5e755eb 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -762,10 +762,7 @@
 		return 0;
 
 	trace_hda_unsol_event(bus, res, res_ex);
-	unsol = bus->unsol;
-	if (!unsol)
-		return 0;
-
+	unsol = &bus->unsol;
 	wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE;
 	unsol->wp = wp;
 
@@ -784,9 +781,8 @@
  */
 static void process_unsol_events(struct work_struct *work)
 {
-	struct hda_bus_unsolicited *unsol =
-		container_of(work, struct hda_bus_unsolicited, work);
-	struct hda_bus *bus = unsol->bus;
+	struct hda_bus *bus = container_of(work, struct hda_bus, unsol.work);
+	struct hda_bus_unsolicited *unsol = &bus->unsol;
 	struct hda_codec *codec;
 	unsigned int rp, caddr, res;
 
@@ -805,27 +801,6 @@
 }
 
 /*
- * initialize unsolicited queue
- */
-static int init_unsol_queue(struct hda_bus *bus)
-{
-	struct hda_bus_unsolicited *unsol;
-
-	if (bus->unsol) /* already initialized */
-		return 0;
-
-	unsol = kzalloc(sizeof(*unsol), GFP_KERNEL);
-	if (!unsol) {
-		dev_err(bus->card->dev, "can't allocate unsolicited queue\n");
-		return -ENOMEM;
-	}
-	INIT_WORK(&unsol->work, process_unsol_events);
-	unsol->bus = bus;
-	bus->unsol = unsol;
-	return 0;
-}
-
-/*
  * destructor
  */
 static void snd_hda_bus_free(struct hda_bus *bus)
@@ -836,7 +811,6 @@
 	WARN_ON(!list_empty(&bus->codec_list));
 	if (bus->workq)
 		flush_workqueue(bus->workq);
-	kfree(bus->unsol);
 	if (bus->ops.private_free)
 		bus->ops.private_free(bus);
 	if (bus->workq)
@@ -861,14 +835,12 @@
 /**
  * snd_hda_bus_new - create a HDA bus
  * @card: the card entry
- * @temp: the template for hda_bus information
  * @busp: the pointer to store the created bus instance
  *
  * Returns 0 if successful, or a negative error code.
  */
 int snd_hda_bus_new(struct snd_card *card,
-			      const struct hda_bus_template *temp,
-			      struct hda_bus **busp)
+		    struct hda_bus **busp)
 {
 	struct hda_bus *bus;
 	int err;
@@ -877,11 +849,6 @@
 		.dev_free = snd_hda_bus_dev_free,
 	};
 
-	if (snd_BUG_ON(!temp))
-		return -EINVAL;
-	if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response))
-		return -EINVAL;
-
 	if (busp)
 		*busp = NULL;
 
@@ -892,15 +859,10 @@
 	}
 
 	bus->card = card;
-	bus->private_data = temp->private_data;
-	bus->pci = temp->pci;
-	bus->modelname = temp->modelname;
-	bus->power_save = temp->power_save;
-	bus->ops = temp->ops;
-
 	mutex_init(&bus->cmd_mutex);
 	mutex_init(&bus->prepare_mutex);
 	INIT_LIST_HEAD(&bus->codec_list);
+	INIT_WORK(&bus->unsol.work, process_unsol_events);
 
 	snprintf(bus->workq_name, sizeof(bus->workq_name),
 		 "hd-audio%d", card->number);
@@ -1702,12 +1664,6 @@
 		return err;
 	}
 
-	if (codec->patch_ops.unsol_event) {
-		err = init_unsol_queue(codec->bus);
-		if (err < 0)
-			return err;
-	}
-
 	/* audio codec should override the mixer name */
 	if (codec->afg || !*codec->bus->card->mixername)
 		snprintf(codec->bus->card->mixername,
@@ -2192,11 +2148,10 @@
 
 static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 			    int direction, int idx, int mask, int val,
-			    bool init_only)
+			    bool init_only, bool cache_only)
 {
 	struct hda_amp_info *info;
 	unsigned int caps;
-	unsigned int cache_only;
 
 	if (snd_BUG_ON(mask & ~0xff))
 		mask &= 0xff;
@@ -2214,7 +2169,7 @@
 		return 0;
 	}
 	info->vol[ch] = val;
-	cache_only = info->head.dirty = codec->cached_write;
+	info->head.dirty |= cache_only;
 	caps = info->amp_caps;
 	mutex_unlock(&codec->hash_mutex);
 	if (!cache_only)
@@ -2238,7 +2193,8 @@
 int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
 			     int direction, int idx, int mask, int val)
 {
-	return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false);
+	return codec_amp_update(codec, nid, ch, direction, idx, mask, val,
+				false, codec->cached_write);
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update);
 
@@ -2285,7 +2241,8 @@
 int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
 			   int dir, int idx, int mask, int val)
 {
-	return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true);
+	return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true,
+				codec->cached_write);
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init);
 
@@ -2427,8 +2384,8 @@
 	maxval = get_amp_max_value(codec, nid, dir, 0);
 	if (val > maxval)
 		val = maxval;
-	return snd_hda_codec_amp_update(codec, nid, ch, dir, idx,
-					HDA_AMP_VOLMASK, val);
+	return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val,
+				false, !hda_codec_is_power_on(codec));
 }
 
 /**
@@ -2478,14 +2435,12 @@
 	long *valp = ucontrol->value.integer.value;
 	int change = 0;
 
-	snd_hda_power_up(codec);
 	if (chs & 1) {
 		change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp);
 		valp++;
 	}
 	if (chs & 2)
 		change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp);
-	snd_hda_power_down(codec);
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put);
@@ -3153,19 +3108,19 @@
 	long *valp = ucontrol->value.integer.value;
 	int change = 0;
 
-	snd_hda_power_up(codec);
 	if (chs & 1) {
-		change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
-						  HDA_AMP_MUTE,
-						  *valp ? 0 : HDA_AMP_MUTE);
+		change = codec_amp_update(codec, nid, 0, dir, idx,
+					  HDA_AMP_MUTE,
+					  *valp ? 0 : HDA_AMP_MUTE, false,
+					  !hda_codec_is_power_on(codec));
 		valp++;
 	}
 	if (chs & 2)
-		change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
-						   HDA_AMP_MUTE,
-						   *valp ? 0 : HDA_AMP_MUTE);
+		change |= codec_amp_update(codec, nid, 1, dir, idx,
+					   HDA_AMP_MUTE,
+					   *valp ? 0 : HDA_AMP_MUTE, false,
+					   !hda_codec_is_power_on(codec));
 	hda_call_check_power_status(codec, nid);
-	snd_hda_power_down(codec);
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 9c8820f..96421a3 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -66,7 +66,6 @@
 struct hda_codec;
 struct hda_pcm;
 struct hda_pcm_stream;
-struct hda_bus_unsolicited;
 
 /* NID type */
 typedef u16 hda_nid_t;
@@ -101,13 +100,14 @@
 #endif
 };
 
-/* template to pass to the bus constructor */
-struct hda_bus_template {
-	void *private_data;
-	struct pci_dev *pci;
-	const char *modelname;
-	int *power_save;
-	struct hda_bus_ops ops;
+/* unsolicited event handler */
+#define HDA_UNSOL_QUEUE_SIZE	64
+struct hda_bus_unsolicited {
+	/* ring buffer */
+	u32 queue[HDA_UNSOL_QUEUE_SIZE * 2];
+	unsigned int rp, wp;
+	/* workqueue */
+	struct work_struct work;
 };
 
 /*
@@ -119,7 +119,6 @@
 struct hda_bus {
 	struct snd_card *card;
 
-	/* copied from template */
 	void *private_data;
 	struct pci_dev *pci;
 	const char *modelname;
@@ -136,7 +135,7 @@
 	struct mutex prepare_mutex;
 
 	/* unsolicited event queue */
-	struct hda_bus_unsolicited *unsol;
+	struct hda_bus_unsolicited unsol;
 	char workq_name[16];
 	struct workqueue_struct *workq;	/* common workqueue for codecs */
 
@@ -420,8 +419,7 @@
 /*
  * constructors
  */
-int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
-		    struct hda_bus **busp);
+int snd_hda_bus_new(struct snd_card *card, struct hda_bus **busp);
 int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 		      struct hda_codec **codecp);
 int snd_hda_codec_configure(struct hda_codec *codec);
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 17c2637..2e73a59 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -30,7 +30,6 @@
 #include <linux/reboot.h>
 #include <sound/core.h>
 #include <sound/initval.h>
-#include "hda_priv.h"
 #include "hda_controller.h"
 
 #define CREATE_TRACE_POINTS
@@ -732,17 +731,32 @@
 			       azx_get_position(chip, azx_dev));
 }
 
-static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream,
-				struct timespec *ts)
+static int azx_get_time_info(struct snd_pcm_substream *substream,
+			struct timespec *system_ts, struct timespec *audio_ts,
+			struct snd_pcm_audio_tstamp_config *audio_tstamp_config,
+			struct snd_pcm_audio_tstamp_report *audio_tstamp_report)
 {
 	struct azx_dev *azx_dev = get_azx_dev(substream);
 	u64 nsec;
 
-	nsec = timecounter_read(&azx_dev->azx_tc);
-	nsec = div_u64(nsec, 3); /* can be optimized */
-	nsec = azx_adjust_codec_delay(substream, nsec);
+	if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) &&
+		(audio_tstamp_config->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK)) {
 
-	*ts = ns_to_timespec(nsec);
+		snd_pcm_gettime(substream->runtime, system_ts);
+
+		nsec = timecounter_read(&azx_dev->azx_tc);
+		nsec = div_u64(nsec, 3); /* can be optimized */
+		if (audio_tstamp_config->report_delay)
+			nsec = azx_adjust_codec_delay(substream, nsec);
+
+		*audio_ts = ns_to_timespec(nsec);
+
+		audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK;
+		audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */
+		audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */
+
+	} else
+		audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT;
 
 	return 0;
 }
@@ -756,7 +770,8 @@
 				 /* SNDRV_PCM_INFO_RESUME |*/
 				 SNDRV_PCM_INFO_PAUSE |
 				 SNDRV_PCM_INFO_SYNC_START |
-				 SNDRV_PCM_INFO_HAS_WALL_CLOCK |
+				 SNDRV_PCM_INFO_HAS_WALL_CLOCK | /* legacy */
+				 SNDRV_PCM_INFO_HAS_LINK_ATIME |
 				 SNDRV_PCM_INFO_NO_PERIOD_WAKEUP),
 	.formats =		SNDRV_PCM_FMTBIT_S16_LE,
 	.rates =		SNDRV_PCM_RATE_48000,
@@ -842,10 +857,12 @@
 		return -EINVAL;
 	}
 
-	/* disable WALLCLOCK timestamps for capture streams
+	/* disable LINK_ATIME timestamps for capture streams
 	   until we figure out how to handle digital inputs */
-	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-		runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK;
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */
+		runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME;
+	}
 
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	azx_dev->substream = substream;
@@ -877,7 +894,7 @@
 	.prepare = azx_pcm_prepare,
 	.trigger = azx_pcm_trigger,
 	.pointer = azx_pcm_pointer,
-	.wall_clock =  azx_get_wallclock_tstamp,
+	.get_time_info =  azx_get_time_info,
 	.mmap = azx_pcm_mmap,
 	.page = snd_pcm_sgbuf_ops_page,
 };
@@ -1676,7 +1693,7 @@
 	int i;
 
 #ifdef CONFIG_PM
-	if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME)
+	if (azx_has_pm_runtime(chip))
 		if (!pm_runtime_active(chip->card->dev))
 			return IRQ_NONE;
 #endif
@@ -1779,7 +1796,7 @@
 {
 	struct azx *chip = bus->private_data;
 
-	if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
+	if (!azx_has_pm_runtime(chip))
 		return;
 
 	if (power_up)
@@ -1810,41 +1827,65 @@
 	return j;
 }
 
-/* Codec initialization */
-int azx_codec_create(struct azx *chip, const char *model,
-		     unsigned int max_slots,
-		     int *power_save_to)
-{
-	struct hda_bus_template bus_temp;
-	int c, codecs, err;
-
-	memset(&bus_temp, 0, sizeof(bus_temp));
-	bus_temp.private_data = chip;
-	bus_temp.modelname = model;
-	bus_temp.pci = chip->pci;
-	bus_temp.ops.command = azx_send_cmd;
-	bus_temp.ops.get_response = azx_get_response;
-	bus_temp.ops.attach_pcm = azx_attach_pcm_stream;
-	bus_temp.ops.bus_reset = azx_bus_reset;
+static struct hda_bus_ops bus_ops = {
+	.command = azx_send_cmd,
+	.get_response = azx_get_response,
+	.attach_pcm = azx_attach_pcm_stream,
+	.bus_reset = azx_bus_reset,
 #ifdef CONFIG_PM
-	bus_temp.power_save = power_save_to;
-	bus_temp.ops.pm_notify = azx_power_notify;
+	.pm_notify = azx_power_notify,
 #endif
 #ifdef CONFIG_SND_HDA_DSP_LOADER
-	bus_temp.ops.load_dsp_prepare = azx_load_dsp_prepare;
-	bus_temp.ops.load_dsp_trigger = azx_load_dsp_trigger;
-	bus_temp.ops.load_dsp_cleanup = azx_load_dsp_cleanup;
+	.load_dsp_prepare = azx_load_dsp_prepare,
+	.load_dsp_trigger = azx_load_dsp_trigger,
+	.load_dsp_cleanup = azx_load_dsp_cleanup,
 #endif
+};
 
-	err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus);
+/* HD-audio bus initialization */
+int azx_bus_create(struct azx *chip, const char *model, int *power_save_to)
+{
+	struct hda_bus *bus;
+	int err;
+
+	err = snd_hda_bus_new(chip->card, &bus);
 	if (err < 0)
 		return err;
 
+	chip->bus = bus;
+	bus->private_data = chip;
+	bus->pci = chip->pci;
+	bus->modelname = model;
+	bus->ops = bus_ops;
+#ifdef CONFIG_PM
+	bus->power_save = power_save_to;
+#endif
+
 	if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) {
 		dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n");
-		chip->bus->needs_damn_long_delay = 1;
+		bus->needs_damn_long_delay = 1;
 	}
 
+	/* AMD chipsets often cause the communication stalls upon certain
+	 * sequence like the pin-detection.  It seems that forcing the synced
+	 * access works around the stall.  Grrr...
+	 */
+	if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) {
+		dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n");
+		bus->sync_write = 1;
+		bus->allow_bus_reset = 1;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(azx_bus_create);
+
+/* Probe codecs */
+int azx_probe_codecs(struct azx *chip, unsigned int max_slots)
+{
+	struct hda_bus *bus = chip->bus;
+	int c, codecs, err;
+
 	codecs = 0;
 	if (!max_slots)
 		max_slots = AZX_DEFAULT_CODECS;
@@ -1872,21 +1913,11 @@
 		}
 	}
 
-	/* AMD chipsets often cause the communication stalls upon certain
-	 * sequence like the pin-detection.  It seems that forcing the synced
-	 * access works around the stall.  Grrr...
-	 */
-	if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) {
-		dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n");
-		chip->bus->sync_write = 1;
-		chip->bus->allow_bus_reset = 1;
-	}
-
 	/* Then create codec instances */
 	for (c = 0; c < max_slots; c++) {
 		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
 			struct hda_codec *codec;
-			err = snd_hda_codec_new(chip->bus, c, &codec);
+			err = snd_hda_codec_new(bus, c, &codec);
 			if (err < 0)
 				continue;
 			codec->jackpoll_interval = get_jackpoll_interval(chip);
@@ -1900,7 +1931,7 @@
 	}
 	return 0;
 }
-EXPORT_SYMBOL_GPL(azx_codec_create);
+EXPORT_SYMBOL_GPL(azx_probe_codecs);
 
 /* configure each codec instance */
 int azx_codec_configure(struct azx *chip)
@@ -1913,13 +1944,6 @@
 }
 EXPORT_SYMBOL_GPL(azx_codec_configure);
 
-/* mixer creation - all stuff is implemented in hda module */
-int azx_mixer_create(struct azx *chip)
-{
-	return snd_hda_build_controls(chip->bus);
-}
-EXPORT_SYMBOL_GPL(azx_mixer_create);
-
 
 static bool is_input_stream(struct azx *chip, unsigned char index)
 {
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index c90d10f..0d09aa6 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -15,10 +15,399 @@
 #ifndef __SOUND_HDA_CONTROLLER_H
 #define __SOUND_HDA_CONTROLLER_H
 
+#include <linux/timecounter.h>
+#include <linux/interrupt.h>
 #include <sound/core.h>
+#include <sound/pcm.h>
 #include <sound/initval.h>
 #include "hda_codec.h"
-#include "hda_priv.h"
+
+/*
+ * registers
+ */
+#define AZX_REG_GCAP			0x00
+#define   AZX_GCAP_64OK		(1 << 0)   /* 64bit address support */
+#define   AZX_GCAP_NSDO		(3 << 1)   /* # of serial data out signals */
+#define   AZX_GCAP_BSS		(31 << 3)  /* # of bidirectional streams */
+#define   AZX_GCAP_ISS		(15 << 8)  /* # of input streams */
+#define   AZX_GCAP_OSS		(15 << 12) /* # of output streams */
+#define AZX_REG_VMIN			0x02
+#define AZX_REG_VMAJ			0x03
+#define AZX_REG_OUTPAY			0x04
+#define AZX_REG_INPAY			0x06
+#define AZX_REG_GCTL			0x08
+#define   AZX_GCTL_RESET	(1 << 0)   /* controller reset */
+#define   AZX_GCTL_FCNTRL	(1 << 1)   /* flush control */
+#define   AZX_GCTL_UNSOL	(1 << 8)   /* accept unsol. response enable */
+#define AZX_REG_WAKEEN			0x0c
+#define AZX_REG_STATESTS		0x0e
+#define AZX_REG_GSTS			0x10
+#define   AZX_GSTS_FSTS		(1 << 1)   /* flush status */
+#define AZX_REG_INTCTL			0x20
+#define AZX_REG_INTSTS			0x24
+#define AZX_REG_WALLCLK			0x30	/* 24Mhz source */
+#define AZX_REG_OLD_SSYNC		0x34	/* SSYNC for old ICH */
+#define AZX_REG_SSYNC			0x38
+#define AZX_REG_CORBLBASE		0x40
+#define AZX_REG_CORBUBASE		0x44
+#define AZX_REG_CORBWP			0x48
+#define AZX_REG_CORBRP			0x4a
+#define   AZX_CORBRP_RST	(1 << 15)  /* read pointer reset */
+#define AZX_REG_CORBCTL			0x4c
+#define   AZX_CORBCTL_RUN	(1 << 1)   /* enable DMA */
+#define   AZX_CORBCTL_CMEIE	(1 << 0)   /* enable memory error irq */
+#define AZX_REG_CORBSTS			0x4d
+#define   AZX_CORBSTS_CMEI	(1 << 0)   /* memory error indication */
+#define AZX_REG_CORBSIZE		0x4e
+
+#define AZX_REG_RIRBLBASE		0x50
+#define AZX_REG_RIRBUBASE		0x54
+#define AZX_REG_RIRBWP			0x58
+#define   AZX_RIRBWP_RST	(1 << 15)  /* write pointer reset */
+#define AZX_REG_RINTCNT			0x5a
+#define AZX_REG_RIRBCTL			0x5c
+#define   AZX_RBCTL_IRQ_EN	(1 << 0)   /* enable IRQ */
+#define   AZX_RBCTL_DMA_EN	(1 << 1)   /* enable DMA */
+#define   AZX_RBCTL_OVERRUN_EN	(1 << 2)   /* enable overrun irq */
+#define AZX_REG_RIRBSTS			0x5d
+#define   AZX_RBSTS_IRQ		(1 << 0)   /* response irq */
+#define   AZX_RBSTS_OVERRUN	(1 << 2)   /* overrun irq */
+#define AZX_REG_RIRBSIZE		0x5e
+
+#define AZX_REG_IC			0x60
+#define AZX_REG_IR			0x64
+#define AZX_REG_IRS			0x68
+#define   AZX_IRS_VALID		(1<<1)
+#define   AZX_IRS_BUSY		(1<<0)
+
+#define AZX_REG_DPLBASE			0x70
+#define AZX_REG_DPUBASE			0x74
+#define   AZX_DPLBASE_ENABLE	0x1	/* Enable position buffer */
+
+/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
+enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
+
+/* stream register offsets from stream base */
+#define AZX_REG_SD_CTL			0x00
+#define AZX_REG_SD_STS			0x03
+#define AZX_REG_SD_LPIB			0x04
+#define AZX_REG_SD_CBL			0x08
+#define AZX_REG_SD_LVI			0x0c
+#define AZX_REG_SD_FIFOW		0x0e
+#define AZX_REG_SD_FIFOSIZE		0x10
+#define AZX_REG_SD_FORMAT		0x12
+#define AZX_REG_SD_BDLPL		0x18
+#define AZX_REG_SD_BDLPU		0x1c
+
+/* PCI space */
+#define AZX_PCIREG_TCSEL		0x44
+
+/*
+ * other constants
+ */
+
+/* max number of fragments - we may use more if allocating more pages for BDL */
+#define BDL_SIZE		4096
+#define AZX_MAX_BDL_ENTRIES	(BDL_SIZE / 16)
+#define AZX_MAX_FRAG		32
+/* max buffer size - no h/w limit, you can increase as you like */
+#define AZX_MAX_BUF_SIZE	(1024*1024*1024)
+
+/* RIRB int mask: overrun[2], response[0] */
+#define RIRB_INT_RESPONSE	0x01
+#define RIRB_INT_OVERRUN	0x04
+#define RIRB_INT_MASK		0x05
+
+/* STATESTS int mask: S3,SD2,SD1,SD0 */
+#define AZX_MAX_CODECS		8
+#define AZX_DEFAULT_CODECS	4
+#define STATESTS_INT_MASK	((1 << AZX_MAX_CODECS) - 1)
+
+/* SD_CTL bits */
+#define SD_CTL_STREAM_RESET	0x01	/* stream reset bit */
+#define SD_CTL_DMA_START	0x02	/* stream DMA start bit */
+#define SD_CTL_STRIPE		(3 << 16)	/* stripe control */
+#define SD_CTL_TRAFFIC_PRIO	(1 << 18)	/* traffic priority */
+#define SD_CTL_DIR		(1 << 19)	/* bi-directional stream */
+#define SD_CTL_STREAM_TAG_MASK	(0xf << 20)
+#define SD_CTL_STREAM_TAG_SHIFT	20
+
+/* SD_CTL and SD_STS */
+#define SD_INT_DESC_ERR		0x10	/* descriptor error interrupt */
+#define SD_INT_FIFO_ERR		0x08	/* FIFO error interrupt */
+#define SD_INT_COMPLETE		0x04	/* completion interrupt */
+#define SD_INT_MASK		(SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\
+				 SD_INT_COMPLETE)
+
+/* SD_STS */
+#define SD_STS_FIFO_READY	0x20	/* FIFO ready */
+
+/* INTCTL and INTSTS */
+#define AZX_INT_ALL_STREAM	0xff	   /* all stream interrupts */
+#define AZX_INT_CTRL_EN	0x40000000 /* controller interrupt enable bit */
+#define AZX_INT_GLOBAL_EN	0x80000000 /* global interrupt enable bit */
+
+/* below are so far hardcoded - should read registers in future */
+#define AZX_MAX_CORB_ENTRIES	256
+#define AZX_MAX_RIRB_ENTRIES	256
+
+/* driver quirks (capabilities) */
+/* bits 0-7 are used for indicating driver type */
+#define AZX_DCAPS_NO_TCSEL	(1 << 8)	/* No Intel TCSEL bit */
+#define AZX_DCAPS_NO_MSI	(1 << 9)	/* No MSI support */
+#define AZX_DCAPS_SNOOP_MASK	(3 << 10)	/* snoop type mask */
+#define AZX_DCAPS_SNOOP_OFF	(1 << 12)	/* snoop default off */
+#define AZX_DCAPS_RIRB_DELAY	(1 << 13)	/* Long delay in read loop */
+#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14)	/* Put a delay before read */
+#define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
+#define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
+#define AZX_DCAPS_POSFIX_VIA	(1 << 17)	/* Use VIACOMBO as default */
+#define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
+#define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
+#define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
+#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21)	/* no buffer size alignment */
+/* 22 unused */
+#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23)	/* BDLE in 4k boundary */
+#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24)	/* Assign devices in reverse order */
+#define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
+#define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
+#define AZX_DCAPS_I915_POWERWELL (1 << 27)	/* HSW i915 powerwell support */
+#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
+#define AZX_DCAPS_NO_MSI64      (1 << 29)	/* Stick to 32-bit MSIs */
+#define AZX_DCAPS_SEPARATE_STREAM_TAG	(1 << 30) /* capture and playback use separate stream tag */
+
+enum {
+	AZX_SNOOP_TYPE_NONE,
+	AZX_SNOOP_TYPE_SCH,
+	AZX_SNOOP_TYPE_ATI,
+	AZX_SNOOP_TYPE_NVIDIA,
+};
+
+/* HD Audio class code */
+#define PCI_CLASS_MULTIMEDIA_HD_AUDIO	0x0403
+
+struct azx_dev {
+	struct snd_dma_buffer bdl; /* BDL buffer */
+	u32 *posbuf;		/* position buffer pointer */
+
+	unsigned int bufsize;	/* size of the play buffer in bytes */
+	unsigned int period_bytes; /* size of the period in bytes */
+	unsigned int frags;	/* number for period in the play buffer */
+	unsigned int fifo_size;	/* FIFO size */
+	unsigned long start_wallclk;	/* start + minimum wallclk */
+	unsigned long period_wallclk;	/* wallclk for period */
+
+	void __iomem *sd_addr;	/* stream descriptor pointer */
+
+	u32 sd_int_sta_mask;	/* stream int status mask */
+
+	/* pcm support */
+	struct snd_pcm_substream *substream;	/* assigned substream,
+						 * set in PCM open
+						 */
+	unsigned int format_val;	/* format value to be set in the
+					 * controller and the codec
+					 */
+	unsigned char stream_tag;	/* assigned stream */
+	unsigned char index;		/* stream index */
+	int assigned_key;		/* last device# key assigned to */
+
+	unsigned int opened:1;
+	unsigned int running:1;
+	unsigned int irq_pending:1;
+	unsigned int prepared:1;
+	unsigned int locked:1;
+	/*
+	 * For VIA:
+	 *  A flag to ensure DMA position is 0
+	 *  when link position is not greater than FIFO size
+	 */
+	unsigned int insufficient:1;
+	unsigned int wc_marked:1;
+	unsigned int no_period_wakeup:1;
+
+	struct timecounter  azx_tc;
+	struct cyclecounter azx_cc;
+
+	int delay_negative_threshold;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+	/* Allows dsp load to have sole access to the playback stream. */
+	struct mutex dsp_mutex;
+#endif
+};
+
+/* CORB/RIRB */
+struct azx_rb {
+	u32 *buf;		/* CORB/RIRB buffer
+				 * Each CORB entry is 4byte, RIRB is 8byte
+				 */
+	dma_addr_t addr;	/* physical address of CORB/RIRB buffer */
+	/* for RIRB */
+	unsigned short rp, wp;	/* read/write pointers */
+	int cmds[AZX_MAX_CODECS];	/* number of pending requests */
+	u32 res[AZX_MAX_CODECS];	/* last read value */
+};
+
+struct azx;
+
+/* Functions to read/write to hda registers. */
+struct hda_controller_ops {
+	/* Register Access */
+	void (*reg_writel)(u32 value, u32 __iomem *addr);
+	u32 (*reg_readl)(u32 __iomem *addr);
+	void (*reg_writew)(u16 value, u16 __iomem *addr);
+	u16 (*reg_readw)(u16 __iomem *addr);
+	void (*reg_writeb)(u8 value, u8 __iomem *addr);
+	u8 (*reg_readb)(u8 __iomem *addr);
+	/* Disable msi if supported, PCI only */
+	int (*disable_msi_reset_irq)(struct azx *);
+	/* Allocation ops */
+	int (*dma_alloc_pages)(struct azx *chip,
+			       int type,
+			       size_t size,
+			       struct snd_dma_buffer *buf);
+	void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf);
+	int (*substream_alloc_pages)(struct azx *chip,
+				     struct snd_pcm_substream *substream,
+				     size_t size);
+	int (*substream_free_pages)(struct azx *chip,
+				    struct snd_pcm_substream *substream);
+	void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream,
+				 struct vm_area_struct *area);
+	/* Check if current position is acceptable */
+	int (*position_check)(struct azx *chip, struct azx_dev *azx_dev);
+};
+
+struct azx_pcm {
+	struct azx *chip;
+	struct snd_pcm *pcm;
+	struct hda_codec *codec;
+	struct hda_pcm_stream *hinfo[2];
+	struct list_head list;
+};
+
+typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *);
+typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos);
+
+struct azx {
+	struct snd_card *card;
+	struct pci_dev *pci;
+	int dev_index;
+
+	/* chip type specific */
+	int driver_type;
+	unsigned int driver_caps;
+	int playback_streams;
+	int playback_index_offset;
+	int capture_streams;
+	int capture_index_offset;
+	int num_streams;
+	const int *jackpoll_ms; /* per-card jack poll interval */
+
+	/* Register interaction. */
+	const struct hda_controller_ops *ops;
+
+	/* position adjustment callbacks */
+	azx_get_pos_callback_t get_position[2];
+	azx_get_delay_callback_t get_delay[2];
+
+	/* pci resources */
+	unsigned long addr;
+	void __iomem *remap_addr;
+	int irq;
+
+	/* locks */
+	spinlock_t reg_lock;
+	struct mutex open_mutex; /* Prevents concurrent open/close operations */
+
+	/* streams (x num_streams) */
+	struct azx_dev *azx_dev;
+
+	/* PCM */
+	struct list_head pcm_list; /* azx_pcm list */
+
+	/* HD codec */
+	unsigned short codec_mask;
+	int  codec_probe_mask; /* copied from probe_mask option */
+	struct hda_bus *bus;
+	unsigned int beep_mode;
+
+	/* CORB/RIRB */
+	struct azx_rb corb;
+	struct azx_rb rirb;
+
+	/* CORB/RIRB and position buffers */
+	struct snd_dma_buffer rb;
+	struct snd_dma_buffer posbuf;
+
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+	const struct firmware *fw;
+#endif
+
+	/* flags */
+	const int *bdl_pos_adj;
+	int poll_count;
+	unsigned int running:1;
+	unsigned int initialized:1;
+	unsigned int single_cmd:1;
+	unsigned int polling_mode:1;
+	unsigned int msi:1;
+	unsigned int probing:1; /* codec probing phase */
+	unsigned int snoop:1;
+	unsigned int align_buffer_size:1;
+	unsigned int region_requested:1;
+	unsigned int disabled:1; /* disabled by VGA-switcher */
+
+	/* for debugging */
+	unsigned int last_cmd[AZX_MAX_CODECS];
+
+	/* reboot notifier (for mysterious hangup problem at power-down) */
+	struct notifier_block reboot_notifier;
+
+#ifdef CONFIG_SND_HDA_DSP_LOADER
+	struct azx_dev saved_azx_dev;
+#endif
+};
+
+#ifdef CONFIG_X86
+#define azx_snoop(chip)		((chip)->snoop)
+#else
+#define azx_snoop(chip)		true
+#endif
+
+/*
+ * macros for easy use
+ */
+
+#define azx_writel(chip, reg, value) \
+	((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg))
+#define azx_readl(chip, reg) \
+	((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg))
+#define azx_writew(chip, reg, value) \
+	((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg))
+#define azx_readw(chip, reg) \
+	((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg))
+#define azx_writeb(chip, reg, value) \
+	((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg))
+#define azx_readb(chip, reg) \
+	((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg))
+
+#define azx_sd_writel(chip, dev, reg, value) \
+	((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg))
+#define azx_sd_readl(chip, dev, reg) \
+	((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg))
+#define azx_sd_writew(chip, dev, reg, value) \
+	((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg))
+#define azx_sd_readw(chip, dev, reg) \
+	((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg))
+#define azx_sd_writeb(chip, dev, reg, value) \
+	((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg))
+#define azx_sd_readb(chip, dev, reg) \
+	((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
+
+#define azx_has_pm_runtime(chip) \
+	(!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME))
 
 /* PCM setup */
 static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream)
@@ -43,11 +432,9 @@
 irqreturn_t azx_interrupt(int irq, void *dev_id);
 
 /* Codec interface */
-int azx_codec_create(struct azx *chip, const char *model,
-		     unsigned int max_slots,
-		     int *power_save_to);
+int azx_bus_create(struct azx *chip, const char *model, int *power_save_to);
+int azx_probe_codecs(struct azx *chip, unsigned int max_slots);
 int azx_codec_configure(struct azx *chip);
-int azx_mixer_create(struct azx *chip);
 int azx_init_stream(struct azx *chip);
 
 void azx_notifier_register(struct azx *chip);
diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c
index 7148945..52a85d8 100644
--- a/sound/pci/hda/hda_i915.c
+++ b/sound/pci/hda/hda_i915.c
@@ -22,7 +22,7 @@
 #include <linux/component.h>
 #include <drm/i915_component.h>
 #include <sound/core.h>
-#include "hda_priv.h"
+#include "hda_controller.h"
 #include "hda_intel.h"
 
 /* Intel HSW/BDW display HDA controller Extended Mode registers.
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4ca3d5d..4614cb1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -62,7 +62,6 @@
 #include <linux/firmware.h>
 #include "hda_codec.h"
 #include "hda_controller.h"
-#include "hda_priv.h"
 #include "hda_intel.h"
 
 /* position fix mode */
@@ -852,7 +851,7 @@
 	if (chip->disabled || hda->init_failed)
 		return 0;
 
-	if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
+	if (!azx_has_pm_runtime(chip))
 		return 0;
 
 	/* enable controller wake up event */
@@ -885,7 +884,7 @@
 	if (chip->disabled || hda->init_failed)
 		return 0;
 
-	if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
+	if (!azx_has_pm_runtime(chip))
 		return 0;
 
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
@@ -928,8 +927,7 @@
 	if (chip->disabled || hda->init_failed)
 		return 0;
 
-	if (!power_save_controller ||
-	    !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
+	if (!power_save_controller || !azx_has_pm_runtime(chip))
 		return -EBUSY;
 
 	return 0;
@@ -1071,8 +1069,7 @@
 	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
 	int i;
 
-	if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME)
-			&& chip->running)
+	if (azx_has_pm_runtime(chip) && chip->running)
 		pm_runtime_get_noresume(&pci->dev);
 
 	azx_del_card_list(chip);
@@ -1896,12 +1893,14 @@
 #endif
 
 	/* create codec instances */
-	err = azx_codec_create(chip, model[dev],
-			       azx_max_codecs[chip->driver_type],
-			       power_save_addr);
-
+	err = azx_bus_create(chip, model[dev], power_save_addr);
 	if (err < 0)
 		goto out_free;
+
+	err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]);
+	if (err < 0)
+		goto out_free;
+
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
 	if (chip->fw) {
 		err = snd_hda_load_patch(chip->bus, chip->fw->size,
@@ -1926,7 +1925,7 @@
 		goto out_free;
 
 	/* create mixer controls */
-	err = azx_mixer_create(chip);
+	err = snd_hda_build_controls(chip->bus);
 	if (err < 0)
 		goto out_free;
 
@@ -1938,7 +1937,7 @@
 	power_down_all_codecs(chip);
 	azx_notifier_register(chip);
 	azx_add_card_list(chip);
-	if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || hda->use_vga_switcheroo)
+	if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo)
 		pm_runtime_put_noidle(&pci->dev);
 
 out_free:
diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h
index 3486118..d5231f7 100644
--- a/sound/pci/hda/hda_intel.h
+++ b/sound/pci/hda/hda_intel.h
@@ -17,7 +17,7 @@
 #define __SOUND_HDA_INTEL_H
 
 #include <drm/i915_component.h>
-#include "hda_priv.h"
+#include "hda_controller.h"
 
 struct hda_intel {
 	struct azx chip;
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 62658f2..49c08a7 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -466,23 +466,6 @@
 			    const struct snd_hda_pin_quirk *pin_quirk,
 			    const struct hda_fixup *fixlist);
 
-
-/*
- * unsolicited event handler
- */
-
-#define HDA_UNSOL_QUEUE_SIZE	64
-
-struct hda_bus_unsolicited {
-	/* ring buffer */
-	u32 queue[HDA_UNSOL_QUEUE_SIZE * 2];
-	unsigned int rp, wp;
-
-	/* workqueue */
-	struct work_struct work;
-	struct hda_bus *bus;
-};
-
 /* helper macros to retrieve pin default-config values */
 #define get_defcfg_connect(cfg) \
 	((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h
deleted file mode 100644
index daf4582..0000000
--- a/sound/pci/hda/hda_priv.h
+++ /dev/null
@@ -1,406 +0,0 @@
-/*
- *  Common defines for the alsa driver code base for HD Audio.
- *
- *  This program is free software; you can redistribute it and/or modify it
- *  under the terms of the GNU General Public License as published by the Free
- *  Software Foundation; either version 2 of the License, or (at your option)
- *  any later version.
- *
- *  This program is distributed in the hope that it will be useful, but WITHOUT
- *  ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
- *  FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
- *  more details.
- */
-
-#ifndef __SOUND_HDA_PRIV_H
-#define __SOUND_HDA_PRIV_H
-
-#include <linux/timecounter.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-
-/*
- * registers
- */
-#define AZX_REG_GCAP			0x00
-#define   AZX_GCAP_64OK		(1 << 0)   /* 64bit address support */
-#define   AZX_GCAP_NSDO		(3 << 1)   /* # of serial data out signals */
-#define   AZX_GCAP_BSS		(31 << 3)  /* # of bidirectional streams */
-#define   AZX_GCAP_ISS		(15 << 8)  /* # of input streams */
-#define   AZX_GCAP_OSS		(15 << 12) /* # of output streams */
-#define AZX_REG_VMIN			0x02
-#define AZX_REG_VMAJ			0x03
-#define AZX_REG_OUTPAY			0x04
-#define AZX_REG_INPAY			0x06
-#define AZX_REG_GCTL			0x08
-#define   AZX_GCTL_RESET	(1 << 0)   /* controller reset */
-#define   AZX_GCTL_FCNTRL	(1 << 1)   /* flush control */
-#define   AZX_GCTL_UNSOL	(1 << 8)   /* accept unsol. response enable */
-#define AZX_REG_WAKEEN			0x0c
-#define AZX_REG_STATESTS		0x0e
-#define AZX_REG_GSTS			0x10
-#define   AZX_GSTS_FSTS		(1 << 1)   /* flush status */
-#define AZX_REG_INTCTL			0x20
-#define AZX_REG_INTSTS			0x24
-#define AZX_REG_WALLCLK			0x30	/* 24Mhz source */
-#define AZX_REG_OLD_SSYNC		0x34	/* SSYNC for old ICH */
-#define AZX_REG_SSYNC			0x38
-#define AZX_REG_CORBLBASE		0x40
-#define AZX_REG_CORBUBASE		0x44
-#define AZX_REG_CORBWP			0x48
-#define AZX_REG_CORBRP			0x4a
-#define   AZX_CORBRP_RST	(1 << 15)  /* read pointer reset */
-#define AZX_REG_CORBCTL			0x4c
-#define   AZX_CORBCTL_RUN	(1 << 1)   /* enable DMA */
-#define   AZX_CORBCTL_CMEIE	(1 << 0)   /* enable memory error irq */
-#define AZX_REG_CORBSTS			0x4d
-#define   AZX_CORBSTS_CMEI	(1 << 0)   /* memory error indication */
-#define AZX_REG_CORBSIZE		0x4e
-
-#define AZX_REG_RIRBLBASE		0x50
-#define AZX_REG_RIRBUBASE		0x54
-#define AZX_REG_RIRBWP			0x58
-#define   AZX_RIRBWP_RST	(1 << 15)  /* write pointer reset */
-#define AZX_REG_RINTCNT			0x5a
-#define AZX_REG_RIRBCTL			0x5c
-#define   AZX_RBCTL_IRQ_EN	(1 << 0)   /* enable IRQ */
-#define   AZX_RBCTL_DMA_EN	(1 << 1)   /* enable DMA */
-#define   AZX_RBCTL_OVERRUN_EN	(1 << 2)   /* enable overrun irq */
-#define AZX_REG_RIRBSTS			0x5d
-#define   AZX_RBSTS_IRQ		(1 << 0)   /* response irq */
-#define   AZX_RBSTS_OVERRUN	(1 << 2)   /* overrun irq */
-#define AZX_REG_RIRBSIZE		0x5e
-
-#define AZX_REG_IC			0x60
-#define AZX_REG_IR			0x64
-#define AZX_REG_IRS			0x68
-#define   AZX_IRS_VALID		(1<<1)
-#define   AZX_IRS_BUSY		(1<<0)
-
-#define AZX_REG_DPLBASE			0x70
-#define AZX_REG_DPUBASE			0x74
-#define   AZX_DPLBASE_ENABLE	0x1	/* Enable position buffer */
-
-/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
-enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
-
-/* stream register offsets from stream base */
-#define AZX_REG_SD_CTL			0x00
-#define AZX_REG_SD_STS			0x03
-#define AZX_REG_SD_LPIB			0x04
-#define AZX_REG_SD_CBL			0x08
-#define AZX_REG_SD_LVI			0x0c
-#define AZX_REG_SD_FIFOW		0x0e
-#define AZX_REG_SD_FIFOSIZE		0x10
-#define AZX_REG_SD_FORMAT		0x12
-#define AZX_REG_SD_BDLPL		0x18
-#define AZX_REG_SD_BDLPU		0x1c
-
-/* PCI space */
-#define AZX_PCIREG_TCSEL		0x44
-
-/*
- * other constants
- */
-
-/* max number of fragments - we may use more if allocating more pages for BDL */
-#define BDL_SIZE		4096
-#define AZX_MAX_BDL_ENTRIES	(BDL_SIZE / 16)
-#define AZX_MAX_FRAG		32
-/* max buffer size - no h/w limit, you can increase as you like */
-#define AZX_MAX_BUF_SIZE	(1024*1024*1024)
-
-/* RIRB int mask: overrun[2], response[0] */
-#define RIRB_INT_RESPONSE	0x01
-#define RIRB_INT_OVERRUN	0x04
-#define RIRB_INT_MASK		0x05
-
-/* STATESTS int mask: S3,SD2,SD1,SD0 */
-#define AZX_MAX_CODECS		8
-#define AZX_DEFAULT_CODECS	4
-#define STATESTS_INT_MASK	((1 << AZX_MAX_CODECS) - 1)
-
-/* SD_CTL bits */
-#define SD_CTL_STREAM_RESET	0x01	/* stream reset bit */
-#define SD_CTL_DMA_START	0x02	/* stream DMA start bit */
-#define SD_CTL_STRIPE		(3 << 16)	/* stripe control */
-#define SD_CTL_TRAFFIC_PRIO	(1 << 18)	/* traffic priority */
-#define SD_CTL_DIR		(1 << 19)	/* bi-directional stream */
-#define SD_CTL_STREAM_TAG_MASK	(0xf << 20)
-#define SD_CTL_STREAM_TAG_SHIFT	20
-
-/* SD_CTL and SD_STS */
-#define SD_INT_DESC_ERR		0x10	/* descriptor error interrupt */
-#define SD_INT_FIFO_ERR		0x08	/* FIFO error interrupt */
-#define SD_INT_COMPLETE		0x04	/* completion interrupt */
-#define SD_INT_MASK		(SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\
-				 SD_INT_COMPLETE)
-
-/* SD_STS */
-#define SD_STS_FIFO_READY	0x20	/* FIFO ready */
-
-/* INTCTL and INTSTS */
-#define AZX_INT_ALL_STREAM	0xff	   /* all stream interrupts */
-#define AZX_INT_CTRL_EN	0x40000000 /* controller interrupt enable bit */
-#define AZX_INT_GLOBAL_EN	0x80000000 /* global interrupt enable bit */
-
-/* below are so far hardcoded - should read registers in future */
-#define AZX_MAX_CORB_ENTRIES	256
-#define AZX_MAX_RIRB_ENTRIES	256
-
-/* driver quirks (capabilities) */
-/* bits 0-7 are used for indicating driver type */
-#define AZX_DCAPS_NO_TCSEL	(1 << 8)	/* No Intel TCSEL bit */
-#define AZX_DCAPS_NO_MSI	(1 << 9)	/* No MSI support */
-#define AZX_DCAPS_SNOOP_MASK	(3 << 10)	/* snoop type mask */
-#define AZX_DCAPS_SNOOP_OFF	(1 << 12)	/* snoop default off */
-#define AZX_DCAPS_RIRB_DELAY	(1 << 13)	/* Long delay in read loop */
-#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14)	/* Put a delay before read */
-#define AZX_DCAPS_CTX_WORKAROUND (1 << 15)	/* X-Fi workaround */
-#define AZX_DCAPS_POSFIX_LPIB	(1 << 16)	/* Use LPIB as default */
-#define AZX_DCAPS_POSFIX_VIA	(1 << 17)	/* Use VIACOMBO as default */
-#define AZX_DCAPS_NO_64BIT	(1 << 18)	/* No 64bit address */
-#define AZX_DCAPS_SYNC_WRITE	(1 << 19)	/* sync each cmd write */
-#define AZX_DCAPS_OLD_SSYNC	(1 << 20)	/* Old SSYNC reg for ICH */
-#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21)	/* no buffer size alignment */
-/* 22 unused */
-#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23)	/* BDLE in 4k boundary */
-#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24)	/* Assign devices in reverse order */
-#define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
-#define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
-#define AZX_DCAPS_I915_POWERWELL (1 << 27)	/* HSW i915 powerwell support */
-#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
-#define AZX_DCAPS_NO_MSI64      (1 << 29)	/* Stick to 32-bit MSIs */
-#define AZX_DCAPS_SEPARATE_STREAM_TAG	(1 << 30) /* capture and playback use separate stream tag */
-
-enum {
-	AZX_SNOOP_TYPE_NONE ,
-	AZX_SNOOP_TYPE_SCH,
-	AZX_SNOOP_TYPE_ATI,
-	AZX_SNOOP_TYPE_NVIDIA,
-};
-
-/* HD Audio class code */
-#define PCI_CLASS_MULTIMEDIA_HD_AUDIO	0x0403
-
-struct azx_dev {
-	struct snd_dma_buffer bdl; /* BDL buffer */
-	u32 *posbuf;		/* position buffer pointer */
-
-	unsigned int bufsize;	/* size of the play buffer in bytes */
-	unsigned int period_bytes; /* size of the period in bytes */
-	unsigned int frags;	/* number for period in the play buffer */
-	unsigned int fifo_size;	/* FIFO size */
-	unsigned long start_wallclk;	/* start + minimum wallclk */
-	unsigned long period_wallclk;	/* wallclk for period */
-
-	void __iomem *sd_addr;	/* stream descriptor pointer */
-
-	u32 sd_int_sta_mask;	/* stream int status mask */
-
-	/* pcm support */
-	struct snd_pcm_substream *substream;	/* assigned substream,
-						 * set in PCM open
-						 */
-	unsigned int format_val;	/* format value to be set in the
-					 * controller and the codec
-					 */
-	unsigned char stream_tag;	/* assigned stream */
-	unsigned char index;		/* stream index */
-	int assigned_key;		/* last device# key assigned to */
-
-	unsigned int opened:1;
-	unsigned int running:1;
-	unsigned int irq_pending:1;
-	unsigned int prepared:1;
-	unsigned int locked:1;
-	/*
-	 * For VIA:
-	 *  A flag to ensure DMA position is 0
-	 *  when link position is not greater than FIFO size
-	 */
-	unsigned int insufficient:1;
-	unsigned int wc_marked:1;
-	unsigned int no_period_wakeup:1;
-
-	struct timecounter  azx_tc;
-	struct cyclecounter azx_cc;
-
-	int delay_negative_threshold;
-
-#ifdef CONFIG_SND_HDA_DSP_LOADER
-	/* Allows dsp load to have sole access to the playback stream. */
-	struct mutex dsp_mutex;
-#endif
-};
-
-/* CORB/RIRB */
-struct azx_rb {
-	u32 *buf;		/* CORB/RIRB buffer
-				 * Each CORB entry is 4byte, RIRB is 8byte
-				 */
-	dma_addr_t addr;	/* physical address of CORB/RIRB buffer */
-	/* for RIRB */
-	unsigned short rp, wp;	/* read/write pointers */
-	int cmds[AZX_MAX_CODECS];	/* number of pending requests */
-	u32 res[AZX_MAX_CODECS];	/* last read value */
-};
-
-struct azx;
-
-/* Functions to read/write to hda registers. */
-struct hda_controller_ops {
-	/* Register Access */
-	void (*reg_writel)(u32 value, u32 __iomem *addr);
-	u32 (*reg_readl)(u32 __iomem *addr);
-	void (*reg_writew)(u16 value, u16 __iomem *addr);
-	u16 (*reg_readw)(u16 __iomem *addr);
-	void (*reg_writeb)(u8 value, u8 __iomem *addr);
-	u8 (*reg_readb)(u8 __iomem *addr);
-	/* Disable msi if supported, PCI only */
-	int (*disable_msi_reset_irq)(struct azx *);
-	/* Allocation ops */
-	int (*dma_alloc_pages)(struct azx *chip,
-			       int type,
-			       size_t size,
-			       struct snd_dma_buffer *buf);
-	void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf);
-	int (*substream_alloc_pages)(struct azx *chip,
-				     struct snd_pcm_substream *substream,
-				     size_t size);
-	int (*substream_free_pages)(struct azx *chip,
-				    struct snd_pcm_substream *substream);
-	void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream,
-				 struct vm_area_struct *area);
-	/* Check if current position is acceptable */
-	int (*position_check)(struct azx *chip, struct azx_dev *azx_dev);
-};
-
-struct azx_pcm {
-	struct azx *chip;
-	struct snd_pcm *pcm;
-	struct hda_codec *codec;
-	struct hda_pcm_stream *hinfo[2];
-	struct list_head list;
-};
-
-typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *);
-typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos);
-
-struct azx {
-	struct snd_card *card;
-	struct pci_dev *pci;
-	int dev_index;
-
-	/* chip type specific */
-	int driver_type;
-	unsigned int driver_caps;
-	int playback_streams;
-	int playback_index_offset;
-	int capture_streams;
-	int capture_index_offset;
-	int num_streams;
-	const int *jackpoll_ms; /* per-card jack poll interval */
-
-	/* Register interaction. */
-	const struct hda_controller_ops *ops;
-
-	/* position adjustment callbacks */
-	azx_get_pos_callback_t get_position[2];
-	azx_get_delay_callback_t get_delay[2];
-
-	/* pci resources */
-	unsigned long addr;
-	void __iomem *remap_addr;
-	int irq;
-
-	/* locks */
-	spinlock_t reg_lock;
-	struct mutex open_mutex; /* Prevents concurrent open/close operations */
-
-	/* streams (x num_streams) */
-	struct azx_dev *azx_dev;
-
-	/* PCM */
-	struct list_head pcm_list; /* azx_pcm list */
-
-	/* HD codec */
-	unsigned short codec_mask;
-	int  codec_probe_mask; /* copied from probe_mask option */
-	struct hda_bus *bus;
-	unsigned int beep_mode;
-
-	/* CORB/RIRB */
-	struct azx_rb corb;
-	struct azx_rb rirb;
-
-	/* CORB/RIRB and position buffers */
-	struct snd_dma_buffer rb;
-	struct snd_dma_buffer posbuf;
-
-#ifdef CONFIG_SND_HDA_PATCH_LOADER
-	const struct firmware *fw;
-#endif
-
-	/* flags */
-	const int *bdl_pos_adj;
-	int poll_count;
-	unsigned int running:1;
-	unsigned int initialized:1;
-	unsigned int single_cmd:1;
-	unsigned int polling_mode:1;
-	unsigned int msi:1;
-	unsigned int probing:1; /* codec probing phase */
-	unsigned int snoop:1;
-	unsigned int align_buffer_size:1;
-	unsigned int region_requested:1;
-	unsigned int disabled:1; /* disabled by VGA-switcher */
-
-	/* for debugging */
-	unsigned int last_cmd[AZX_MAX_CODECS];
-
-	/* reboot notifier (for mysterious hangup problem at power-down) */
-	struct notifier_block reboot_notifier;
-
-#ifdef CONFIG_SND_HDA_DSP_LOADER
-	struct azx_dev saved_azx_dev;
-#endif
-};
-
-#ifdef CONFIG_X86
-#define azx_snoop(chip)		((chip)->snoop)
-#else
-#define azx_snoop(chip)		true
-#endif
-
-/*
- * macros for easy use
- */
-
-#define azx_writel(chip, reg, value) \
-	((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg))
-#define azx_readl(chip, reg) \
-	((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg))
-#define azx_writew(chip, reg, value) \
-	((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg))
-#define azx_readw(chip, reg) \
-	((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg))
-#define azx_writeb(chip, reg, value) \
-	((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg))
-#define azx_readb(chip, reg) \
-	((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg))
-
-#define azx_sd_writel(chip, dev, reg, value) \
-	((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg))
-#define azx_sd_readl(chip, dev, reg) \
-	((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg))
-#define azx_sd_writew(chip, dev, reg, value) \
-	((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg))
-#define azx_sd_readw(chip, dev, reg) \
-	((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg))
-#define azx_sd_writeb(chip, dev, reg, value) \
-	((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg))
-#define azx_sd_readb(chip, dev, reg) \
-	((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
-
-#endif /* __SOUND_HDA_PRIV_H */
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 375e94f..1bd7a9e 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -37,7 +37,6 @@
 
 #include "hda_codec.h"
 #include "hda_controller.h"
-#include "hda_priv.h"
 
 /* Defines for Nvidia Tegra HDA support */
 #define HDA_BAR0           0x8000
@@ -503,7 +502,11 @@
 		goto out_free;
 
 	/* create codec instances */
-	err = azx_codec_create(chip, NULL, 0, &power_save);
+	err = azx_bus_create(chip, NULL, &power_save);
+	if (err < 0)
+		goto out_free;
+
+	err = azx_probe_codecs(chip, 0);
 	if (err < 0)
 		goto out_free;
 
@@ -517,7 +520,7 @@
 		goto out_free;
 
 	/* create mixer controls */
-	err = azx_mixer_create(chip);
+	err = snd_hda_build_controls(chip->bus);
 	if (err < 0)
 		goto out_free;
 
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index ca67f89..cb666c7 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6043,23 +6043,30 @@
 	.mask = 0
 };
 
-static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
+static int snd_hdspm_open(struct snd_pcm_substream *substream)
 {
 	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	spin_lock_irq(&hdspm->lock);
-
 	snd_pcm_set_sync(substream);
+	runtime->hw = (playback) ? snd_hdspm_playback_subinfo :
+		snd_hdspm_capture_subinfo;
 
+	if (playback) {
+		if (hdspm->capture_substream == NULL)
+			hdspm_stop_audio(hdspm);
 
-	runtime->hw = snd_hdspm_playback_subinfo;
+		hdspm->playback_pid = current->pid;
+		hdspm->playback_substream = substream;
+	} else {
+		if (hdspm->playback_substream == NULL)
+			hdspm_stop_audio(hdspm);
 
-	if (hdspm->capture_substream == NULL)
-		hdspm_stop_audio(hdspm);
-
-	hdspm->playback_pid = current->pid;
-	hdspm->playback_substream = substream;
+		hdspm->capture_pid = current->pid;
+		hdspm->capture_substream = substream;
+	}
 
 	spin_unlock_irq(&hdspm->lock);
 
@@ -6094,108 +6101,42 @@
 				&hdspm_hw_constraints_aes32_sample_rates);
 	} else {
 		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				snd_hdspm_hw_rule_rate_out_channels, hdspm,
+				(playback ?
+				 snd_hdspm_hw_rule_rate_out_channels :
+				 snd_hdspm_hw_rule_rate_in_channels), hdspm,
 				SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 	}
 
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_out_channels, hdspm,
+			(playback ? snd_hdspm_hw_rule_out_channels :
+			 snd_hdspm_hw_rule_in_channels), hdspm,
 			SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_out_channels_rate, hdspm,
+			(playback ? snd_hdspm_hw_rule_out_channels_rate :
+			 snd_hdspm_hw_rule_in_channels_rate), hdspm,
 			SNDRV_PCM_HW_PARAM_RATE, -1);
 
 	return 0;
 }
 
-static int snd_hdspm_playback_release(struct snd_pcm_substream *substream)
+static int snd_hdspm_release(struct snd_pcm_substream *substream)
 {
 	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+	bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	spin_lock_irq(&hdspm->lock);
 
-	hdspm->playback_pid = -1;
-	hdspm->playback_substream = NULL;
-
-	spin_unlock_irq(&hdspm->lock);
-
-	return 0;
-}
-
-
-static int snd_hdspm_capture_open(struct snd_pcm_substream *substream)
-{
-	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	spin_lock_irq(&hdspm->lock);
-	snd_pcm_set_sync(substream);
-	runtime->hw = snd_hdspm_capture_subinfo;
-
-	if (hdspm->playback_substream == NULL)
-		hdspm_stop_audio(hdspm);
-
-	hdspm->capture_pid = current->pid;
-	hdspm->capture_substream = substream;
-
-	spin_unlock_irq(&hdspm->lock);
-
-	snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
-	snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
-
-	switch (hdspm->io_type) {
-	case AIO:
-	case RayDAT:
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-					     32, 4096);
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
-					     16384, 16384);
-		break;
-
-	default:
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-					     64, 8192);
-		snd_pcm_hw_constraint_minmax(runtime,
-					     SNDRV_PCM_HW_PARAM_PERIODS,
-					     2, 2);
-		break;
-	}
-
-	if (AES32 == hdspm->io_type) {
-		runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
-		snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				&hdspm_hw_constraints_aes32_sample_rates);
+	if (playback) {
+		hdspm->playback_pid = -1;
+		hdspm->playback_substream = NULL;
 	} else {
-		snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				snd_hdspm_hw_rule_rate_in_channels, hdspm,
-				SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+		hdspm->capture_pid = -1;
+		hdspm->capture_substream = NULL;
 	}
 
-	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_in_channels, hdspm,
-			SNDRV_PCM_HW_PARAM_CHANNELS, -1);
-
-	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-			snd_hdspm_hw_rule_in_channels_rate, hdspm,
-			SNDRV_PCM_HW_PARAM_RATE, -1);
-
-	return 0;
-}
-
-static int snd_hdspm_capture_release(struct snd_pcm_substream *substream)
-{
-	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
-
-	spin_lock_irq(&hdspm->lock);
-
-	hdspm->capture_pid = -1;
-	hdspm->capture_substream = NULL;
-
 	spin_unlock_irq(&hdspm->lock);
+
 	return 0;
 }
 
@@ -6413,21 +6354,9 @@
 	return 0;
 }
 
-static struct snd_pcm_ops snd_hdspm_playback_ops = {
-	.open = snd_hdspm_playback_open,
-	.close = snd_hdspm_playback_release,
-	.ioctl = snd_hdspm_ioctl,
-	.hw_params = snd_hdspm_hw_params,
-	.hw_free = snd_hdspm_hw_free,
-	.prepare = snd_hdspm_prepare,
-	.trigger = snd_hdspm_trigger,
-	.pointer = snd_hdspm_hw_pointer,
-	.page = snd_pcm_sgbuf_ops_page,
-};
-
-static struct snd_pcm_ops snd_hdspm_capture_ops = {
-	.open = snd_hdspm_capture_open,
-	.close = snd_hdspm_capture_release,
+static struct snd_pcm_ops snd_hdspm_ops = {
+	.open = snd_hdspm_open,
+	.close = snd_hdspm_release,
 	.ioctl = snd_hdspm_ioctl,
 	.hw_params = snd_hdspm_hw_params,
 	.hw_free = snd_hdspm_hw_free,
@@ -6521,9 +6450,9 @@
 	strcpy(pcm->name, hdspm->card_name);
 
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
-			&snd_hdspm_playback_ops);
+			&snd_hdspm_ops);
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
-			&snd_hdspm_capture_ops);
+			&snd_hdspm_ops);
 
 	pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
 
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index fb0b7e8b..841d059 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -187,6 +187,94 @@
 	return IRQ_HANDLED;
 }
 
+/*
+ * When the bit clock is input, limit the maximum rate according to the
+ * Serial Clock Ratio Considerations section from the SSC documentation:
+ *
+ *   The Transmitter and the Receiver can be programmed to operate
+ *   with the clock signals provided on either the TK or RK pins.
+ *   This allows the SSC to support many slave-mode data transfers.
+ *   In this case, the maximum clock speed allowed on the RK pin is:
+ *   - Peripheral clock divided by 2 if Receiver Frame Synchro is input
+ *   - Peripheral clock divided by 3 if Receiver Frame Synchro is output
+ *   In addition, the maximum clock speed allowed on the TK pin is:
+ *   - Peripheral clock divided by 6 if Transmit Frame Synchro is input
+ *   - Peripheral clock divided by 2 if Transmit Frame Synchro is output
+ *
+ * When the bit clock is output, limit the rate according to the
+ * SSC divider restrictions.
+ */
+static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params,
+				  struct snd_pcm_hw_rule *rule)
+{
+	struct atmel_ssc_info *ssc_p = rule->private;
+	struct ssc_device *ssc = ssc_p->ssc;
+	struct snd_interval *i = hw_param_interval(params, rule->var);
+	struct snd_interval t;
+	struct snd_ratnum r = {
+		.den_min = 1,
+		.den_max = 4095,
+		.den_step = 1,
+	};
+	unsigned int num = 0, den = 0;
+	int frame_size;
+	int mck_div = 2;
+	int ret;
+
+	frame_size = snd_soc_params_to_frame_size(params);
+	if (frame_size < 0)
+		return frame_size;
+
+	switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFS:
+		if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE)
+		    && ssc->clk_from_rk_pin)
+			/* Receiver Frame Synchro (i.e. capture)
+			 * is output (format is _CFS) and the RK pin
+			 * is used for input (format is _CBM_).
+			 */
+			mck_div = 3;
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK)
+		    && !ssc->clk_from_rk_pin)
+			/* Transmit Frame Synchro (i.e. playback)
+			 * is input (format is _CFM) and the TK pin
+			 * is used for input (format _CBM_ but not
+			 * using the RK pin).
+			 */
+			mck_div = 6;
+		break;
+	}
+
+	switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		r.num = ssc_p->mck_rate / mck_div / frame_size;
+
+		ret = snd_interval_ratnum(i, 1, &r, &num, &den);
+		if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) {
+			params->rate_num = num;
+			params->rate_den = den;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBM_CFM:
+		t.min = 8000;
+		t.max = ssc_p->mck_rate / mck_div / frame_size;
+		t.openmin = t.openmax = 0;
+		t.integer = 0;
+		ret = snd_interval_refine(i, &t);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
 
 /*-------------------------------------------------------------------------*\
  * DAI functions
@@ -200,6 +288,7 @@
 	struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
 	struct atmel_pcm_dma_params *dma_params;
 	int dir, dir_mask;
+	int ret;
 
 	pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
 		ssc_readl(ssc_p->ssc->regs, SR));
@@ -207,6 +296,7 @@
 	/* Enable PMC peripheral clock for this SSC */
 	pr_debug("atmel_ssc_dai: Starting clock\n");
 	clk_enable(ssc_p->ssc->clk);
+	ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk);
 
 	/* Reset the SSC to keep it at a clean status */
 	ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
@@ -219,6 +309,17 @@
 		dir_mask = SSC_DIR_MASK_CAPTURE;
 	}
 
+	ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+				  SNDRV_PCM_HW_PARAM_RATE,
+				  atmel_ssc_hw_rule_rate,
+				  ssc_p,
+				  SNDRV_PCM_HW_PARAM_FRAME_BITS,
+				  SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+	if (ret < 0) {
+		dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret);
+		return ret;
+	}
+
 	dma_params = &ssc_dma_params[dai->id][dir];
 	dma_params->ssc = ssc_p->ssc;
 	dma_params->substream = substream;
@@ -783,8 +884,6 @@
 #  define atmel_ssc_resume	NULL
 #endif /* CONFIG_PM */
 
-#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000)
-
 #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
 			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
@@ -804,12 +903,16 @@
 		.playback = {
 			.channels_min = 1,
 			.channels_max = 2,
-			.rates = ATMEL_SSC_RATES,
+			.rates = SNDRV_PCM_RATE_CONTINUOUS,
+			.rate_min = 8000,
+			.rate_max = 384000,
 			.formats = ATMEL_SSC_FORMATS,},
 		.capture = {
 			.channels_min = 1,
 			.channels_max = 2,
-			.rates = ATMEL_SSC_RATES,
+			.rates = SNDRV_PCM_RATE_CONTINUOUS,
+			.rate_min = 8000,
+			.rate_max = 384000,
 			.formats = ATMEL_SSC_FORMATS,},
 		.ops = &atmel_ssc_dai_ops,
 };
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index b1f08d5..80b1538 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -115,6 +115,7 @@
 	unsigned short rcmr_period;
 	struct atmel_pcm_dma_params *dma_params[2];
 	struct atmel_ssc_state ssc_state;
+	unsigned long mck_rate;
 };
 
 int atmel_ssc_set_audio(int ssc_id);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index ea9f0e3..0bddd92 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -141,7 +141,8 @@
 	select SND_SOC_WM8770 if SPI_MASTER
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8782
-	select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8804_I2C if I2C
+	select SND_SOC_WM8804_SPI if SPI_MASTER
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
 	select SND_SOC_WM8904 if I2C
@@ -744,8 +745,19 @@
 	tristate
 
 config SND_SOC_WM8804
-	tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver"
-	depends on SND_SOC_I2C_AND_SPI
+	tristate
+
+config SND_SOC_WM8804_I2C
+	tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C"
+	depends on I2C
+	select SND_SOC_WM8804
+	select REGMAP_I2C
+
+config SND_SOC_WM8804_SPI
+	tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI"
+	depends on SPI_MASTER
+	select SND_SOC_WM8804
+	select REGMAP_SPI
 
 config SND_SOC_WM8900
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 69b8666..7acb6c1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -145,6 +145,8 @@
 snd-soc-wm8776-objs := wm8776.o
 snd-soc-wm8782-objs := wm8782.o
 snd-soc-wm8804-objs := wm8804.o
+snd-soc-wm8804-i2c-objs := wm8804-i2c.o
+snd-soc-wm8804-spi-objs := wm8804-spi.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
 snd-soc-wm8904-objs := wm8904.o
@@ -323,6 +325,8 @@
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
 obj-$(CONFIG_SND_SOC_WM8782)	+= snd-soc-wm8782.o
 obj-$(CONFIG_SND_SOC_WM8804)	+= snd-soc-wm8804.o
+obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o
+obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8904)	+= snd-soc-wm8904.o
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index 70ab357..7ad8e15 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -938,22 +938,15 @@
 		adau1977->dvdd_reg = NULL;
 	}
 
-	adau1977->reset_gpio = devm_gpiod_get(dev, "reset");
-	if (IS_ERR(adau1977->reset_gpio)) {
-		ret = PTR_ERR(adau1977->reset_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return PTR_ERR(adau1977->reset_gpio);
-		adau1977->reset_gpio = NULL;
-	}
+	adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset",
+						       GPIOD_OUT_LOW);
+	if (IS_ERR(adau1977->reset_gpio))
+		return PTR_ERR(adau1977->reset_gpio);
 
 	dev_set_drvdata(dev, adau1977);
 
-	if (adau1977->reset_gpio) {
-		ret = gpiod_direction_output(adau1977->reset_gpio, 0);
-		if (ret)
-			return ret;
+	if (adau1977->reset_gpio)
 		ndelay(100);
-	}
 
 	ret = adau1977_power_enable(adau1977);
 	if (ret)
diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c
index f2b8aad..60598b2 100644
--- a/sound/soc/codecs/cs35l32.c
+++ b/sound/soc/codecs/cs35l32.c
@@ -437,20 +437,13 @@
 	}
 
 	/* Reset the Device */
-	cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev,
-		"reset-gpios");
-	if (IS_ERR(cs35l32->reset_gpio)) {
-		ret = PTR_ERR(cs35l32->reset_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
+	cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
+		"reset", GPIOD_OUT_LOW);
+	if (IS_ERR(cs35l32->reset_gpio))
+		return PTR_ERR(cs35l32->reset_gpio);
 
-		cs35l32->reset_gpio = NULL;
-	} else {
-		ret = gpiod_direction_output(cs35l32->reset_gpio, 0);
-		if (ret)
-			return ret;
+	if (cs35l32->reset_gpio)
 		gpiod_set_value_cansleep(cs35l32->reset_gpio, 1);
-	}
 
 	/* initialize codec */
 	ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, &reg);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index ce60868..cac48dd 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -605,21 +605,14 @@
 		return ret;
 	}
 
-	cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev,
-		"reset-gpios");
-	if (IS_ERR(cs4265->reset_gpio)) {
-		ret = PTR_ERR(cs4265->reset_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
+	cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev,
+		"reset", GPIOD_OUT_LOW);
+	if (IS_ERR(cs4265->reset_gpio))
+		return PTR_ERR(cs4265->reset_gpio);
 
-		cs4265->reset_gpio = NULL;
-	} else {
-		ret = gpiod_direction_output(cs4265->reset_gpio, 0);
-		if (ret)
-			return ret;
+	if (cs4265->reset_gpio) {
 		mdelay(1);
 		gpiod_set_value_cansleep(cs4265->reset_gpio, 1);
-
 	}
 
 	i2c_set_clientdata(i2c_client, cs4265);
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index e9e6efb..bf3e933 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -26,8 +26,6 @@
 #include <sound/soc-dai.h>
 #include <sound/soc-dapm.h>
 
-#define DRV_NAME "max98357a"
-
 static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
 		int cmd, struct snd_soc_dai *dai)
 {
@@ -87,9 +85,9 @@
 };
 
 static struct snd_soc_dai_driver max98357a_dai_driver = {
-	.name = DRV_NAME,
+	.name = "HiFi",
 	.playback = {
-		.stream_name	= DRV_NAME "-playback",
+		.stream_name	= "HiFi Playback",
 		.formats	= SNDRV_PCM_FMTBIT_S16 |
 					SNDRV_PCM_FMTBIT_S24 |
 					SNDRV_PCM_FMTBIT_S32,
@@ -127,7 +125,7 @@
 
 #ifdef CONFIG_OF
 static const struct of_device_id max98357a_device_id[] = {
-	{ .compatible = "maxim," DRV_NAME, },
+	{ .compatible = "maxim,max98357a" },
 	{}
 };
 MODULE_DEVICE_TABLE(of, max98357a_device_id);
@@ -135,7 +133,7 @@
 
 static struct platform_driver max98357a_platform_driver = {
 	.driver = {
-		.name = DRV_NAME,
+		.name = "max98357a",
 		.of_match_table = of_match_ptr(max98357a_device_id),
 	},
 	.probe	= max98357a_platform_probe,
@@ -145,4 +143,3 @@
 
 MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver");
 MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 9974f20..4b5f1fe 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -54,6 +54,9 @@
 	int pll_d;
 	int pll_p;
 	unsigned long real_pll;
+	unsigned long overclock_pll;
+	unsigned long overclock_dac;
+	unsigned long overclock_dsp;
 };
 
 /*
@@ -224,6 +227,90 @@
 	}
 }
 
+static int pcm512x_overclock_pll_get(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = pcm512x->overclock_pll;
+	return 0;
+}
+
+static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+	switch (codec->dapm.bias_level) {
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_STANDBY:
+		break;
+	default:
+		return -EBUSY;
+	}
+
+	pcm512x->overclock_pll = ucontrol->value.integer.value[0];
+	return 0;
+}
+
+static int pcm512x_overclock_dsp_get(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = pcm512x->overclock_dsp;
+	return 0;
+}
+
+static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+	switch (codec->dapm.bias_level) {
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_STANDBY:
+		break;
+	default:
+		return -EBUSY;
+	}
+
+	pcm512x->overclock_dsp = ucontrol->value.integer.value[0];
+	return 0;
+}
+
+static int pcm512x_overclock_dac_get(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = pcm512x->overclock_dac;
+	return 0;
+}
+
+static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
+
+	switch (codec->dapm.bias_level) {
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_STANDBY:
+		break;
+	default:
+		return -EBUSY;
+	}
+
+	pcm512x->overclock_dac = ucontrol->value.integer.value[0];
+	return 0;
+}
+
 static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1);
 static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0);
 static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
@@ -328,6 +415,13 @@
 SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus),
 SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf),
 SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds),
+
+SOC_SINGLE_EXT("Max Overclock PLL", SND_SOC_NOPM, 0, 20, 0,
+	       pcm512x_overclock_pll_get, pcm512x_overclock_pll_put),
+SOC_SINGLE_EXT("Max Overclock DSP", SND_SOC_NOPM, 0, 40, 0,
+	       pcm512x_overclock_dsp_get, pcm512x_overclock_dsp_put),
+SOC_SINGLE_EXT("Max Overclock DAC", SND_SOC_NOPM, 0, 40, 0,
+	       pcm512x_overclock_dac_get, pcm512x_overclock_dac_put),
 };
 
 static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = {
@@ -346,6 +440,45 @@
 	{ "OUTR", NULL, "DACR" },
 };
 
+static unsigned long pcm512x_pll_max(struct pcm512x_priv *pcm512x)
+{
+	return 25000000 + 25000000 * pcm512x->overclock_pll / 100;
+}
+
+static unsigned long pcm512x_dsp_max(struct pcm512x_priv *pcm512x)
+{
+	return 50000000 + 50000000 * pcm512x->overclock_dsp / 100;
+}
+
+static unsigned long pcm512x_dac_max(struct pcm512x_priv *pcm512x,
+				     unsigned long rate)
+{
+	return rate + rate * pcm512x->overclock_dac / 100;
+}
+
+static unsigned long pcm512x_sck_max(struct pcm512x_priv *pcm512x)
+{
+	if (!pcm512x->pll_out)
+		return 25000000;
+	return pcm512x_pll_max(pcm512x);
+}
+
+static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x,
+					unsigned long dac_rate)
+{
+	/*
+	 * If the DAC is not actually overclocked, use the good old
+	 * NCP target rate...
+	 */
+	if (dac_rate <= 6144000)
+		return 1536000;
+	/*
+	 * ...but if the DAC is in fact overclocked, bump the NCP target
+	 * rate to get the recommended dividers even when overclocking.
+	 */
+	return pcm512x_dac_max(pcm512x, 1536000);
+}
+
 static const u32 pcm512x_dai_rates[] = {
 	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
 	88200, 96000, 176400, 192000, 384000,
@@ -359,6 +492,7 @@
 static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params,
 				struct snd_pcm_hw_rule *rule)
 {
+	struct pcm512x_priv *pcm512x = rule->private;
 	struct snd_interval ranges[2];
 	int frame_size;
 
@@ -377,7 +511,7 @@
 		 */
 		memset(ranges, 0, sizeof(ranges));
 		ranges[0].min = 8000;
-		ranges[0].max = 25000000 / frame_size / 2;
+		ranges[0].max = pcm512x_sck_max(pcm512x) / frame_size / 2;
 		ranges[1].min = DIV_ROUND_UP(16000000, frame_size);
 		ranges[1].max = 384000;
 		break;
@@ -408,7 +542,7 @@
 		return snd_pcm_hw_rule_add(substream->runtime, 0,
 					   SNDRV_PCM_HW_PARAM_RATE,
 					   pcm512x_hw_rule_rate,
-					   NULL,
+					   pcm512x,
 					   SNDRV_PCM_HW_PARAM_FRAME_BITS,
 					   SNDRV_PCM_HW_PARAM_CHANNELS, -1);
 
@@ -517,6 +651,8 @@
 				      unsigned long bclk_rate)
 {
 	struct device *dev = dai->dev;
+	struct snd_soc_codec *codec = dai->codec;
+	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 	unsigned long sck_rate;
 	int pow2;
 
@@ -527,9 +663,10 @@
 	 * as many factors of 2 as possible, as that makes it easier
 	 * to find a fast DAC rate
 	 */
-	pow2 = 1 << fls((25000000 - 16000000) / bclk_rate);
+	pow2 = 1 << fls((pcm512x_pll_max(pcm512x) - 16000000) / bclk_rate);
 	for (; pow2; pow2 >>= 1) {
-		sck_rate = rounddown(25000000, bclk_rate * pow2);
+		sck_rate = rounddown(pcm512x_pll_max(pcm512x),
+				     bclk_rate * pow2);
 		if (sck_rate >= 16000000)
 			break;
 	}
@@ -678,7 +815,7 @@
 		return 0; /* futile, quit early */
 
 	/* run DAC no faster than 6144000 Hz */
-	for (dac_rate = rounddown(6144000, osr_rate);
+	for (dac_rate = rounddown(pcm512x_dac_max(pcm512x, 6144000), osr_rate);
 	     dac_rate;
 	     dac_rate -= osr_rate) {
 
@@ -805,7 +942,7 @@
 	osr_rate = 16 * sample_rate;
 
 	/* run DSP no faster than 50 MHz */
-	dsp_div = mck_rate > 50000000 ? 2 : 1;
+	dsp_div = mck_rate > pcm512x_dsp_max(pcm512x) ? 2 : 1;
 
 	dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate);
 	if (dac_rate) {
@@ -836,7 +973,8 @@
 		dacsrc_rate = pllin_rate;
 	} else {
 		/* run DAC no faster than 6144000 Hz */
-		unsigned long dac_mul = 6144000 / osr_rate;
+		unsigned long dac_mul = pcm512x_dac_max(pcm512x, 6144000)
+			/ osr_rate;
 		unsigned long sck_mul = sck_rate / osr_rate;
 
 		for (; dac_mul; dac_mul--) {
@@ -863,28 +1001,30 @@
 		dacsrc_rate = sck_rate;
 	}
 
+	osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate);
+	if (osr_div > 128) {
+		dev_err(dev, "Failed to find OSR divider\n");
+		return -EINVAL;
+	}
+
 	dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate);
 	if (dac_div > 128) {
 		dev_err(dev, "Failed to find DAC divider\n");
 		return -EINVAL;
 	}
+	dac_rate = dacsrc_rate / dac_div;
 
-	ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000);
-	if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) {
+	ncp_div = DIV_ROUND_CLOSEST(dac_rate,
+				    pcm512x_ncp_target(pcm512x, dac_rate));
+	if (ncp_div > 128 || dac_rate / ncp_div > 2048000) {
 		/* run NCP no faster than 2048000 Hz, but why? */
-		ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000);
+		ncp_div = DIV_ROUND_UP(dac_rate, 2048000);
 		if (ncp_div > 128) {
 			dev_err(dev, "Failed to find NCP divider\n");
 			return -EINVAL;
 		}
 	}
 
-	osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate);
-	if (osr_div > 128) {
-		dev_err(dev, "Failed to find OSR divider\n");
-		return -EINVAL;
-	}
-
 	idac = mck_rate / (dsp_div * sample_rate);
 
 	ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1);
@@ -937,11 +1077,11 @@
 		return ret;
 	}
 
-	if (sample_rate <= 48000)
+	if (sample_rate <= pcm512x_dac_max(pcm512x, 48000))
 		fssp = PCM512x_FSSP_48KHZ;
-	else if (sample_rate <= 96000)
+	else if (sample_rate <= pcm512x_dac_max(pcm512x, 96000))
 		fssp = PCM512x_FSSP_96KHZ;
-	else if (sample_rate <= 192000)
+	else if (sample_rate <= pcm512x_dac_max(pcm512x, 192000))
 		fssp = PCM512x_FSSP_192KHZ;
 	else
 		fssp = PCM512x_FSSP_384KHZ;
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index f374840..16723b1 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -395,9 +395,20 @@
 
 	rt286->jack = jack;
 
-	/* Send an initial empty report */
-	snd_soc_jack_report(rt286->jack, 0,
-		SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+	if (jack) {
+		/* enable IRQ */
+		if (rt286->jack->status | SND_JACK_HEADPHONE)
+			snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1");
+		regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2);
+		/* Send an initial empty report */
+		snd_soc_jack_report(rt286->jack, rt286->jack->status,
+			SND_JACK_MICROPHONE | SND_JACK_HEADPHONE);
+	} else {
+		/* disable IRQ */
+		regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0);
+		snd_soc_dapm_disable_pin(&codec->dapm, "LDO1");
+	}
+	snd_soc_dapm_sync(&codec->dapm);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index 21f8e18..0a67adb 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -1950,17 +1950,20 @@
 };
 
 enum {
+	RT5670_DMIC1_DISABLED,
 	RT5670_DMIC_DATA_GPIO6,
 	RT5670_DMIC_DATA_IN2P,
 	RT5670_DMIC_DATA_GPIO7,
 };
 
 enum {
+	RT5670_DMIC2_DISABLED,
 	RT5670_DMIC_DATA_GPIO8,
 	RT5670_DMIC_DATA_IN3N,
 };
 
 enum {
+	RT5670_DMIC3_DISABLED,
 	RT5670_DMIC_DATA_GPIO9,
 	RT5670_DMIC_DATA_GPIO10,
 	RT5670_DMIC_DATA_GPIO5,
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index fb9c20e..c2a6e40 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -718,11 +718,24 @@
 			RT5677_LDO1_SEL_MASK, 0x0);
 		regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2,
 			RT5677_PWR_LDO1, RT5677_PWR_LDO1);
-		regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1,
-			RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC);
-		regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2,
-			RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK,
-			RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS);
+		switch (rt5677->type) {
+		case RT5677:
+			regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1,
+				RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC);
+			regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2,
+				RT5677_PLL2_PR_SRC_MASK |
+				RT5677_DSP_CLK_SRC_MASK,
+				RT5677_PLL2_PR_SRC_MCLK2 |
+				RT5677_DSP_CLK_SRC_BYPASS);
+			break;
+		case RT5676:
+			regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2,
+				RT5677_DSP_CLK_SRC_MASK,
+				RT5677_DSP_CLK_SRC_BYPASS);
+			break;
+		default:
+			break;
+		}
 		regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff);
 		regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd);
 		rt5677_set_dsp_mode(codec, true);
@@ -4500,10 +4513,10 @@
 	if (!rt5677->dsp_vad_en) {
 		regcache_cache_only(rt5677->regmap, true);
 		regcache_mark_dirty(rt5677->regmap);
-	}
 
-	if (gpio_is_valid(rt5677->pow_ldo2))
-		gpio_set_value_cansleep(rt5677->pow_ldo2, 0);
+		if (gpio_is_valid(rt5677->pow_ldo2))
+			gpio_set_value_cansleep(rt5677->pow_ldo2, 0);
+	}
 
 	return 0;
 }
@@ -4512,12 +4525,12 @@
 {
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
-	if (gpio_is_valid(rt5677->pow_ldo2)) {
-		gpio_set_value_cansleep(rt5677->pow_ldo2, 1);
-		msleep(10);
-	}
-
 	if (!rt5677->dsp_vad_en) {
+		if (gpio_is_valid(rt5677->pow_ldo2)) {
+			gpio_set_value_cansleep(rt5677->pow_ldo2, 1);
+			msleep(10);
+		}
+
 		regcache_cache_only(rt5677->regmap, false);
 		regcache_sync(rt5677->regmap);
 	}
@@ -4733,7 +4746,8 @@
 };
 
 static const struct i2c_device_id rt5677_i2c_id[] = {
-	{ "rt5677", 0 },
+	{ "rt5677", RT5677 },
+	{ "rt5676", RT5676 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id);
@@ -4850,6 +4864,8 @@
 
 	i2c_set_clientdata(i2c, rt5677);
 
+	rt5677->type = id->driver_data;
+
 	if (pdata)
 		rt5677->pdata = *pdata;
 
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index c0a625f..07df96b 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -1665,6 +1665,11 @@
 	RT5677_IRQ_JD3,
 };
 
+enum rt5677_type {
+	RT5677,
+	RT5676,
+};
+
 struct rt5677_priv {
 	struct snd_soc_codec *codec;
 	struct rt5677_platform_data pdata;
@@ -1681,6 +1686,7 @@
 	int pll_in;
 	int pll_out;
 	int pow_ldo2; /* POW_LDO2 pin */
+	enum rt5677_type type;
 #ifdef CONFIG_GPIOLIB
 	struct gpio_chip gpio_chip;
 #endif
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 47b257e..1e5d264 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -783,19 +783,21 @@
 	snd_soc_write(codec, SN95031_BTNCTRL2, 0x01);
 }
 
-static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack)
+static int sn95031_get_headset_state(struct snd_soc_codec *codec,
+	struct snd_soc_jack *mfld_jack)
 {
-	int micbias = sn95031_get_mic_bias(mfld_jack->codec);
+	int micbias = sn95031_get_mic_bias(codec);
 
 	int jack_type = snd_soc_jack_get_type(mfld_jack, micbias);
 
 	pr_debug("jack type detected = %d\n", jack_type);
 	if (jack_type == SND_JACK_HEADSET)
-		sn95031_enable_jack_btn(mfld_jack->codec);
+		sn95031_enable_jack_btn(codec);
 	return jack_type;
 }
 
-void sn95031_jack_detection(struct mfld_jack_data *jack_data)
+void sn95031_jack_detection(struct snd_soc_codec *codec,
+	struct mfld_jack_data *jack_data)
 {
 	unsigned int status;
 	unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET;
@@ -809,11 +811,11 @@
 		status = SND_JACK_HEADSET | SND_JACK_BTN_1;
 	} else if (jack_data->intr_id & 0x4) {
 		pr_debug("headset or headphones inserted\n");
-		status = sn95031_get_headset_state(jack_data->mfld_jack);
+		status = sn95031_get_headset_state(codec, jack_data->mfld_jack);
 	} else if (jack_data->intr_id & 0x8) {
 		pr_debug("headset or headphones removed\n");
 		status = 0;
-		sn95031_disable_jack_btn(jack_data->mfld_jack->codec);
+		sn95031_disable_jack_btn(codec);
 	} else {
 		pr_err("unidentified interrupt\n");
 		return;
diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h
index 20376d2..7651fe4 100644
--- a/sound/soc/codecs/sn95031.h
+++ b/sound/soc/codecs/sn95031.h
@@ -127,6 +127,7 @@
 	struct snd_soc_jack *mfld_jack;
 };
 
-extern void sn95031_jack_detection(struct mfld_jack_data *jack_data);
+extern void sn95031_jack_detection(struct snd_soc_codec *codec,
+	struct mfld_jack_data *jack_data);
 
 #endif
diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c
index bda2ee1..669e322 100644
--- a/sound/soc/codecs/sta350.c
+++ b/sound/soc/codecs/sta350.c
@@ -1213,27 +1213,15 @@
 #endif
 
 	/* GPIOs */
-	sta350->gpiod_nreset = devm_gpiod_get(dev, "reset");
-	if (IS_ERR(sta350->gpiod_nreset)) {
-		ret = PTR_ERR(sta350->gpiod_nreset);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
+	sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset",
+						       GPIOD_OUT_LOW);
+	if (IS_ERR(sta350->gpiod_nreset))
+		return PTR_ERR(sta350->gpiod_nreset);
 
-		sta350->gpiod_nreset = NULL;
-	} else {
-		gpiod_direction_output(sta350->gpiod_nreset, 0);
-	}
-
-	sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down");
-	if (IS_ERR(sta350->gpiod_power_down)) {
-		ret = PTR_ERR(sta350->gpiod_power_down);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
-
-		sta350->gpiod_power_down = NULL;
-	} else {
-		gpiod_direction_output(sta350->gpiod_power_down, 0);
-	}
+	sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down",
+						  GPIOD_OUT_LOW);
+	if (IS_ERR(sta350->gpiod_power_down))
+		return PTR_ERR(sta350->gpiod_power_down);
 
 	/* regulators */
 	for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++)
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index ae23acd..dfb4ff5 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -485,16 +485,9 @@
 	if (data == NULL)
 		return -ENOMEM;
 
-	data->enable_gpio = devm_gpiod_get(dev, "enable");
-	if (IS_ERR(data->enable_gpio)) {
-		ret = PTR_ERR(data->enable_gpio);
-		if (ret != -ENOENT && ret != -ENOSYS)
-			return ret;
-
-		data->enable_gpio = NULL;
-	} else {
-		gpiod_direction_output(data->enable_gpio, 0);
-	}
+	data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW);
+	if (IS_ERR(data->enable_gpio))
+		return PTR_ERR(data->enable_gpio);
 
 	data->tas2552_client = client;
 	data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config);
diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c
new file mode 100644
index 0000000..5bd4af2
--- /dev/null
+++ b/sound/soc/codecs/wm8804-i2c.c
@@ -0,0 +1,64 @@
+/*
+ * wm8804-i2c.c  --  WM8804 S/PDIF transceiver driver - I2C
+ *
+ * Copyright 2015 Cirrus Logic Inc
+ *
+ * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+
+#include "wm8804.h"
+
+static int wm8804_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct regmap *regmap;
+
+	regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	return wm8804_probe(&i2c->dev, regmap);
+}
+
+static int wm8804_i2c_remove(struct i2c_client *i2c)
+{
+	wm8804_remove(&i2c->dev);
+	return 0;
+}
+
+static const struct i2c_device_id wm8804_i2c_id[] = {
+	{ "wm8804", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id);
+
+static const struct of_device_id wm8804_of_match[] = {
+	{ .compatible = "wlf,wm8804", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8804_of_match);
+
+static struct i2c_driver wm8804_i2c_driver = {
+	.driver = {
+		.name = "wm8804",
+		.owner = THIS_MODULE,
+		.of_match_table = wm8804_of_match,
+	},
+	.probe = wm8804_i2c_probe,
+	.remove = wm8804_i2c_remove,
+	.id_table = wm8804_i2c_id
+};
+
+module_i2c_driver(wm8804_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC WM8804 driver - I2C");
+MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c
new file mode 100644
index 0000000..287e11e
--- /dev/null
+++ b/sound/soc/codecs/wm8804-spi.c
@@ -0,0 +1,56 @@
+/*
+ * wm8804-spi.c  --  WM8804 S/PDIF transceiver driver - SPI
+ *
+ * Copyright 2015 Cirrus Logic Inc
+ *
+ * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+
+#include "wm8804.h"
+
+static int wm8804_spi_probe(struct spi_device *spi)
+{
+	struct regmap *regmap;
+
+	regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	return wm8804_probe(&spi->dev, regmap);
+}
+
+static int wm8804_spi_remove(struct spi_device *spi)
+{
+	wm8804_remove(&spi->dev);
+	return 0;
+}
+
+static const struct of_device_id wm8804_of_match[] = {
+	{ .compatible = "wlf,wm8804", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8804_of_match);
+
+static struct spi_driver wm8804_spi_driver = {
+	.driver = {
+		.name = "wm8804",
+		.owner = THIS_MODULE,
+		.of_match_table = wm8804_of_match,
+	},
+	.probe = wm8804_spi_probe,
+	.remove = wm8804_spi_remove
+};
+
+module_spi_driver(wm8804_spi_driver);
+
+MODULE_DESCRIPTION("ASoC WM8804 driver - SPI");
+MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index b2b0e68..1bd4ace 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -15,10 +15,7 @@
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
-#include <linux/i2c.h>
 #include <linux/of_device.h>
-#include <linux/spi/spi.h>
-#include <linux/regmap.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -185,9 +182,9 @@
 	}
 }
 
-static int wm8804_reset(struct snd_soc_codec *codec)
+static int wm8804_reset(struct wm8804_priv *wm8804)
 {
-	return snd_soc_write(codec, WM8804_RST_DEVID1, 0x0);
+	return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0);
 }
 
 static int wm8804_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
@@ -518,100 +515,6 @@
 	return 0;
 }
 
-static int wm8804_remove(struct snd_soc_codec *codec)
-{
-	struct wm8804_priv *wm8804;
-	int i;
-
-	wm8804 = snd_soc_codec_get_drvdata(codec);
-
-	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i)
-		regulator_unregister_notifier(wm8804->supplies[i].consumer,
-					      &wm8804->disable_nb[i]);
-	return 0;
-}
-
-static int wm8804_probe(struct snd_soc_codec *codec)
-{
-	struct wm8804_priv *wm8804;
-	int i, id1, id2, ret;
-
-	wm8804 = snd_soc_codec_get_drvdata(codec);
-
-	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++)
-		wm8804->supplies[i].supply = wm8804_supply_names[i];
-
-	ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies),
-				 wm8804->supplies);
-	if (ret) {
-		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
-		return ret;
-	}
-
-	wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0;
-	wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1;
-
-	/* This should really be moved into the regulator core */
-	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) {
-		ret = regulator_register_notifier(wm8804->supplies[i].consumer,
-						  &wm8804->disable_nb[i]);
-		if (ret != 0) {
-			dev_err(codec->dev,
-				"Failed to register regulator notifier: %d\n",
-				ret);
-		}
-	}
-
-	ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies),
-				    wm8804->supplies);
-	if (ret) {
-		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
-		return ret;
-	}
-
-	id1 = snd_soc_read(codec, WM8804_RST_DEVID1);
-	if (id1 < 0) {
-		dev_err(codec->dev, "Failed to read device ID: %d\n", id1);
-		ret = id1;
-		goto err_reg_enable;
-	}
-
-	id2 = snd_soc_read(codec, WM8804_DEVID2);
-	if (id2 < 0) {
-		dev_err(codec->dev, "Failed to read device ID: %d\n", id2);
-		ret = id2;
-		goto err_reg_enable;
-	}
-
-	id2 = (id2 << 8) | id1;
-
-	if (id2 != 0x8805) {
-		dev_err(codec->dev, "Invalid device ID: %#x\n", id2);
-		ret = -EINVAL;
-		goto err_reg_enable;
-	}
-
-	ret = snd_soc_read(codec, WM8804_DEVREV);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read device revision: %d\n",
-			ret);
-		goto err_reg_enable;
-	}
-	dev_info(codec->dev, "revision %c\n", ret + 'A');
-
-	ret = wm8804_reset(codec);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
-		goto err_reg_enable;
-	}
-
-	return 0;
-
-err_reg_enable:
-	regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies);
-	return ret;
-}
-
 static const struct snd_soc_dai_ops wm8804_dai_ops = {
 	.hw_params = wm8804_hw_params,
 	.set_fmt = wm8804_set_fmt,
@@ -649,8 +552,6 @@
 };
 
 static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = {
-	.probe = wm8804_probe,
-	.remove = wm8804_remove,
 	.set_bias_level = wm8804_set_bias_level,
 	.idle_bias_off = true,
 
@@ -658,13 +559,7 @@
 	.num_controls = ARRAY_SIZE(wm8804_snd_controls),
 };
 
-static const struct of_device_id wm8804_of_match[] = {
-	{ .compatible = "wlf,wm8804", },
-	{ }
-};
-MODULE_DEVICE_TABLE(of, wm8804_of_match);
-
-static const struct regmap_config wm8804_regmap_config = {
+const struct regmap_config wm8804_regmap_config = {
 	.reg_bits = 8,
 	.val_bits = 8,
 
@@ -675,128 +570,110 @@
 	.reg_defaults = wm8804_reg_defaults,
 	.num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults),
 };
+EXPORT_SYMBOL_GPL(wm8804_regmap_config);
 
-#if defined(CONFIG_SPI_MASTER)
-static int wm8804_spi_probe(struct spi_device *spi)
+int wm8804_probe(struct device *dev, struct regmap *regmap)
 {
 	struct wm8804_priv *wm8804;
-	int ret;
+	unsigned int id1, id2;
+	int i, ret;
 
-	wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL);
+	wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL);
 	if (!wm8804)
 		return -ENOMEM;
 
-	wm8804->regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config);
-	if (IS_ERR(wm8804->regmap)) {
-		ret = PTR_ERR(wm8804->regmap);
-		return ret;
-	}
+	dev_set_drvdata(dev, wm8804);
 
-	spi_set_drvdata(spi, wm8804);
+	wm8804->regmap = regmap;
 
-	ret = snd_soc_register_codec(&spi->dev,
-				     &soc_codec_dev_wm8804, &wm8804_dai, 1);
+	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++)
+		wm8804->supplies[i].supply = wm8804_supply_names[i];
 
-	return ret;
-}
-
-static int wm8804_spi_remove(struct spi_device *spi)
-{
-	snd_soc_unregister_codec(&spi->dev);
-	return 0;
-}
-
-static struct spi_driver wm8804_spi_driver = {
-	.driver = {
-		.name = "wm8804",
-		.owner = THIS_MODULE,
-		.of_match_table = wm8804_of_match,
-	},
-	.probe = wm8804_spi_probe,
-	.remove = wm8804_spi_remove
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-static int wm8804_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
-{
-	struct wm8804_priv *wm8804;
-	int ret;
-
-	wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL);
-	if (!wm8804)
-		return -ENOMEM;
-
-	wm8804->regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config);
-	if (IS_ERR(wm8804->regmap)) {
-		ret = PTR_ERR(wm8804->regmap);
-		return ret;
-	}
-
-	i2c_set_clientdata(i2c, wm8804);
-
-	ret = snd_soc_register_codec(&i2c->dev,
-				     &soc_codec_dev_wm8804, &wm8804_dai, 1);
-	return ret;
-}
-
-static int wm8804_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_codec(&i2c->dev);
-	return 0;
-}
-
-static const struct i2c_device_id wm8804_i2c_id[] = {
-	{ "wm8804", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id);
-
-static struct i2c_driver wm8804_i2c_driver = {
-	.driver = {
-		.name = "wm8804",
-		.owner = THIS_MODULE,
-		.of_match_table = wm8804_of_match,
-	},
-	.probe = wm8804_i2c_probe,
-	.remove = wm8804_i2c_remove,
-	.id_table = wm8804_i2c_id
-};
-#endif
-
-static int __init wm8804_modinit(void)
-{
-	int ret = 0;
-
-#if IS_ENABLED(CONFIG_I2C)
-	ret = i2c_add_driver(&wm8804_i2c_driver);
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8804->supplies),
+				      wm8804->supplies);
 	if (ret) {
-		printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n",
-		       ret);
+		dev_err(dev, "Failed to request supplies: %d\n", ret);
+		return ret;
 	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	ret = spi_register_driver(&wm8804_spi_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register wm8804 SPI driver: %d\n",
-		       ret);
+
+	wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0;
+	wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1;
+
+	/* This should really be moved into the regulator core */
+	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) {
+		ret = regulator_register_notifier(wm8804->supplies[i].consumer,
+						  &wm8804->disable_nb[i]);
+		if (ret != 0) {
+			dev_err(dev,
+				"Failed to register regulator notifier: %d\n",
+				ret);
+		}
 	}
-#endif
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies),
+				    wm8804->supplies);
+	if (ret) {
+		dev_err(dev, "Failed to enable supplies: %d\n", ret);
+		goto err_reg_enable;
+	}
+
+	ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1);
+	if (ret < 0) {
+		dev_err(dev, "Failed to read device ID: %d\n", ret);
+		goto err_reg_enable;
+	}
+
+	ret = regmap_read(regmap, WM8804_DEVID2, &id2);
+	if (ret < 0) {
+		dev_err(dev, "Failed to read device ID: %d\n", ret);
+		goto err_reg_enable;
+	}
+
+	id2 = (id2 << 8) | id1;
+
+	if (id2 != 0x8805) {
+		dev_err(dev, "Invalid device ID: %#x\n", id2);
+		ret = -EINVAL;
+		goto err_reg_enable;
+	}
+
+	ret = regmap_read(regmap, WM8804_DEVREV, &id1);
+	if (ret < 0) {
+		dev_err(dev, "Failed to read device revision: %d\n",
+			ret);
+		goto err_reg_enable;
+	}
+	dev_info(dev, "revision %c\n", id1 + 'A');
+
+	ret = wm8804_reset(wm8804);
+	if (ret < 0) {
+		dev_err(dev, "Failed to issue reset: %d\n", ret);
+		goto err_reg_enable;
+	}
+
+	return snd_soc_register_codec(dev, &soc_codec_dev_wm8804,
+				      &wm8804_dai, 1);
+
+err_reg_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies);
 	return ret;
 }
-module_init(wm8804_modinit);
+EXPORT_SYMBOL_GPL(wm8804_probe);
 
-static void __exit wm8804_exit(void)
+void wm8804_remove(struct device *dev)
 {
-#if IS_ENABLED(CONFIG_I2C)
-	i2c_del_driver(&wm8804_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8804_spi_driver);
-#endif
+	struct wm8804_priv *wm8804;
+	int i;
+
+	wm8804 = dev_get_drvdata(dev);
+
+	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i)
+		regulator_unregister_notifier(wm8804->supplies[i].consumer,
+					      &wm8804->disable_nb[i]);
+
+	snd_soc_unregister_codec(dev);
 }
-module_exit(wm8804_exit);
+EXPORT_SYMBOL_GPL(wm8804_remove);
 
 MODULE_DESCRIPTION("ASoC WM8804 driver");
 MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h
index e72d4f4..a39a256 100644
--- a/sound/soc/codecs/wm8804.h
+++ b/sound/soc/codecs/wm8804.h
@@ -13,6 +13,8 @@
 #ifndef _WM8804_H
 #define _WM8804_H
 
+#include <linux/regmap.h>
+
 /*
  * Register values.
  */
@@ -62,4 +64,9 @@
 #define WM8804_MCLKDIV_256FS			0
 #define WM8804_MCLKDIV_128FS			1
 
+extern const struct regmap_config wm8804_regmap_config;
+
+int wm8804_probe(struct device *dev, struct regmap *regmap);
+void wm8804_remove(struct device *dev);
+
 #endif  /* _WM8804_H */
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ff67b33..d01c209 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -420,10 +420,9 @@
 
 	memcpy(ctl->cache, p, ctl->len);
 
-	if (!ctl->enabled) {
-		ctl->set = 1;
+	ctl->set = 1;
+	if (!ctl->enabled)
 		return 0;
-	}
 
 	return wm_coeff_write_control(kcontrol, p, ctl->len);
 }
@@ -1185,7 +1184,6 @@
 	int ret, pos, blocks, type, offset, reg;
 	char *file;
 	struct wm_adsp_buf *buf;
-	int tmp;
 
 	file = kzalloc(PAGE_SIZE, GFP_KERNEL);
 	if (file == NULL)
@@ -1335,12 +1333,7 @@
 			}
 		}
 
-		tmp = le32_to_cpu(blk->len) % 4;
-		if (tmp)
-			pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk);
-		else
-			pos += le32_to_cpu(blk->len) + sizeof(*blk);
-
+		pos += (le32_to_cpu(blk->len) + sizeof(*blk) + 3) & ~0x03;
 		blocks++;
 	}
 
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 2b81ca4..3736d9a 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,14 +1,16 @@
 config SND_DAVINCI_SOC
-	tristate "SoC Audio for TI DAVINCI"
+	tristate
 	depends on ARCH_DAVINCI
+	select SND_EDMA_SOC
 
 config SND_EDMA_SOC
-	tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)"
-	depends on SOC_AM33XX || SOC_AM43XX
+	tristate "SoC Audio for Texas Instruments chips using eDMA"
+	depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	help
 	  Say Y or M here if you want audio support for TI SoC which uses eDMA.
 	  The following line of SoCs are supported by this platform driver:
+	  - daVinci devices
 	  - AM335x
 	  - AM437x/AM438x
 
@@ -17,7 +19,7 @@
 
 config SND_DAVINCI_SOC_MCASP
 	tristate "Multichannel Audio Serial Port (McASP) support"
-	depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC
+	depends on SND_OMAP_SOC || SND_EDMA_SOC
 	help
 	  Say Y or M here if you want to have support for McASP IP found in
 	  various Texas Instruments SoCs like:
@@ -45,7 +47,7 @@
 
 config SND_DAVINCI_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
-	depends on SND_DAVINCI_SOC && I2C
+	depends on SND_EDMA_SOC && I2C
 	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM
 	select SND_DAVINCI_SOC_GENERIC_EVM
 	help
@@ -73,7 +75,7 @@
 
 config  SND_DM6467_SOC_EVM
 	tristate "SoC Audio support for DaVinci DM6467 EVM"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM && I2C
+	depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C
 	select SND_DAVINCI_SOC_GENERIC_EVM
 	select SND_SOC_SPDIF
 
@@ -82,7 +84,7 @@
 
 config  SND_DA830_SOC_EVM
 	tristate "SoC Audio support for DA830/OMAP-L137 EVM"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM && I2C
+	depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C
 	select SND_DAVINCI_SOC_GENERIC_EVM
 
 	help
@@ -91,7 +93,7 @@
 
 config  SND_DA850_SOC_EVM
 	tristate "SoC Audio support for DA850/OMAP-L138 EVM"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM && I2C
+	depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C
 	select SND_DAVINCI_SOC_GENERIC_EVM
 	help
 	  Say Y if you want to add support for SoC audio on TI
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index 09bf2ba..f883933 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -1,11 +1,9 @@
 # DAVINCI Platform Support
-snd-soc-davinci-objs := davinci-pcm.o
 snd-soc-edma-objs := edma-pcm.o
 snd-soc-davinci-i2s-objs := davinci-i2s.o
 snd-soc-davinci-mcasp-objs:= davinci-mcasp.o
 snd-soc-davinci-vcif-objs:= davinci-vcif.o
 
-obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
 obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o
 obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 15fb28f..56cb4d9 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -23,8 +23,9 @@
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
-#include "davinci-pcm.h"
+#include "edma-pcm.h"
 #include "davinci-i2s.h"
 
 
@@ -122,7 +123,8 @@
 
 struct davinci_mcbsp_dev {
 	struct device *dev;
-	struct davinci_pcm_dma_params	dma_params[2];
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+	int dma_request[2];
 	void __iomem			*base;
 #define MOD_DSP_A	0
 #define MOD_DSP_B	1
@@ -419,8 +421,6 @@
 				 struct snd_soc_dai *dai)
 {
 	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
-	struct davinci_pcm_dma_params *dma_params =
-					&dev->dma_params[substream->stream];
 	struct snd_interval *i = NULL;
 	int mcbsp_word_length, master;
 	unsigned int rcr, xcr, srgr, clk_div, freq, framesize;
@@ -532,8 +532,6 @@
 			return -EINVAL;
 		}
 	}
-	dma_params->acnt = dma_params->data_type = data_type[fmt];
-	dma_params->fifo_level = 0;
 	mcbsp_word_length = asp_word_length[fmt];
 
 	switch (master) {
@@ -600,15 +598,6 @@
 	return ret;
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
-{
-	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
-
-	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
-	return 0;
-}
-
 static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
@@ -620,7 +609,6 @@
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = {
-	.startup	= davinci_i2s_startup,
 	.shutdown	= davinci_i2s_shutdown,
 	.prepare	= davinci_i2s_prepare,
 	.trigger	= davinci_i2s_trigger,
@@ -630,7 +618,18 @@
 
 };
 
+static int davinci_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+
+	return 0;
+}
+
 static struct snd_soc_dai_driver davinci_i2s_dai = {
+	.probe = davinci_i2s_dai_probe,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -651,11 +650,9 @@
 
 static int davinci_i2s_probe(struct platform_device *pdev)
 {
-	struct snd_platform_data *pdata = pdev->dev.platform_data;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea, *res;
-	enum dma_event_q asp_chan_q = EVENTQ_0;
-	enum dma_event_q ram_chan_q = EVENTQ_1;
+	int *dma;
 	int ret;
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -676,22 +673,6 @@
 			   GFP_KERNEL);
 	if (!dev)
 		return -ENOMEM;
-	if (pdata) {
-		dev->enable_channel_combine = pdata->enable_channel_combine;
-		dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size =
-			pdata->sram_size_playback;
-		dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size =
-			pdata->sram_size_capture;
-		dev->clk_input_pin = pdata->clk_input_pin;
-		dev->i2s_accurate_sck = pdata->i2s_accurate_sck;
-		asp_chan_q = pdata->asp_chan_q;
-		ram_chan_q = pdata->ram_chan_q;
-	}
-
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q	= asp_chan_q;
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q	= ram_chan_q;
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q	= asp_chan_q;
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q	= ram_chan_q;
 
 	dev->clk = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(dev->clk))
@@ -705,10 +686,10 @@
 		goto err_release_clk;
 	}
 
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
+	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
 	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
 
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
+	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
 	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
 
 	/* first TX, then RX */
@@ -718,7 +699,9 @@
 		ret = -ENXIO;
 		goto err_release_clk;
 	}
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
+	dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
+	*dma = res->start;
+	dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma;
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
@@ -726,9 +709,11 @@
 		ret = -ENXIO;
 		goto err_release_clk;
 	}
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
-	dev->dev = &pdev->dev;
+	dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
+	*dma = res->start;
+	dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma;
 
+	dev->dev = &pdev->dev;
 	dev_set_drvdata(&pdev->dev, dev);
 
 	ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component,
@@ -736,7 +721,7 @@
 	if (ret != 0)
 		goto err_release_clk;
 
-	ret = davinci_soc_platform_register(&pdev->dev);
+	ret = edma_pcm_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
 		goto err_unregister_component;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index de3b155..0c88299 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -26,6 +26,7 @@
 #include <linux/of.h>
 #include <linux/of_platform.h>
 #include <linux/of_device.h>
+#include <linux/platform_data/davinci_asp.h>
 
 #include <sound/asoundef.h>
 #include <sound/core.h>
@@ -36,7 +37,6 @@
 #include <sound/dmaengine_pcm.h>
 #include <sound/omap-pcm.h>
 
-#include "davinci-pcm.h"
 #include "edma-pcm.h"
 #include "davinci-mcasp.h"
 
@@ -65,7 +65,6 @@
 };
 
 struct davinci_mcasp {
-	struct davinci_pcm_dma_params dma_params[2];
 	struct snd_dmaengine_dai_dma_data dma_data[2];
 	void __iomem *base;
 	u32 fifo_base;
@@ -82,6 +81,7 @@
 	u16	bclk_lrclk_ratio;
 	int	streams;
 	u32	irq_request[2];
+	int	dma_request[2];
 
 	int	sysclk_freq;
 	bool	bclk_master;
@@ -441,6 +441,18 @@
 		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
 		mcasp->bclk_master = 1;
 		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		/* codec is clock slave and frame master */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
+		mcasp->bclk_master = 1;
+		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		/* codec is clock master and frame slave */
 		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
@@ -631,7 +643,6 @@
 static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 				 int period_words, int channels)
 {
-	struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[stream];
 	struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream];
 	int i;
 	u8 tx_ser = 0;
@@ -699,10 +710,8 @@
 			 * For example if three serializers are enabled the DMA
 			 * need to transfer three words per DMA request.
 			 */
-			dma_params->fifo_level = active_serializers;
 			dma_data->maxburst = active_serializers;
 		} else {
-			dma_params->fifo_level = 0;
 			dma_data->maxburst = 0;
 		}
 		return 0;
@@ -734,7 +743,6 @@
 	/* Configure the burst size for platform drivers */
 	if (numevt == 1)
 		numevt = 0;
-	dma_params->fifo_level = numevt;
 	dma_data->maxburst = numevt;
 
 	return 0;
@@ -860,8 +868,6 @@
 					struct snd_soc_dai *cpu_dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
-	struct davinci_pcm_dma_params *dma_params =
-					&mcasp->dma_params[substream->stream];
 	int word_length;
 	int channels = params_channels(params);
 	int period_size = params_period_size(params);
@@ -902,31 +908,26 @@
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U8:
 	case SNDRV_PCM_FORMAT_S8:
-		dma_params->data_type = 1;
 		word_length = 8;
 		break;
 
 	case SNDRV_PCM_FORMAT_U16_LE:
 	case SNDRV_PCM_FORMAT_S16_LE:
-		dma_params->data_type = 2;
 		word_length = 16;
 		break;
 
 	case SNDRV_PCM_FORMAT_U24_3LE:
 	case SNDRV_PCM_FORMAT_S24_3LE:
-		dma_params->data_type = 3;
 		word_length = 24;
 		break;
 
 	case SNDRV_PCM_FORMAT_U24_LE:
 	case SNDRV_PCM_FORMAT_S24_LE:
-		dma_params->data_type = 4;
 		word_length = 24;
 		break;
 
 	case SNDRV_PCM_FORMAT_U32_LE:
 	case SNDRV_PCM_FORMAT_S32_LE:
-		dma_params->data_type = 4;
 		word_length = 32;
 		break;
 
@@ -935,11 +936,6 @@
 		return -EINVAL;
 	}
 
-	if (mcasp->version == MCASP_VERSION_2 && !dma_params->fifo_level)
-		dma_params->acnt = 4;
-	else
-		dma_params->acnt = dma_params->data_type;
-
 	davinci_config_channel_size(mcasp, word_length);
 
 	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
@@ -1043,17 +1039,8 @@
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 
-	if (mcasp->version >= MCASP_VERSION_3) {
-		/* Using dmaengine PCM */
-		dai->playback_dma_data =
-				&mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
-		dai->capture_dma_data =
-				&mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
-	} else {
-		/* Using davinci-pcm */
-		dai->playback_dma_data = mcasp->dma_params;
-		dai->capture_dma_data = mcasp->dma_params;
-	}
+	dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
 
 	return 0;
 }
@@ -1172,28 +1159,24 @@
 static struct davinci_mcasp_pdata dm646x_mcasp_pdata = {
 	.tx_dma_offset = 0x400,
 	.rx_dma_offset = 0x400,
-	.asp_chan_q = EVENTQ_0,
 	.version = MCASP_VERSION_1,
 };
 
 static struct davinci_mcasp_pdata da830_mcasp_pdata = {
 	.tx_dma_offset = 0x2000,
 	.rx_dma_offset = 0x2000,
-	.asp_chan_q = EVENTQ_0,
 	.version = MCASP_VERSION_2,
 };
 
 static struct davinci_mcasp_pdata am33xx_mcasp_pdata = {
 	.tx_dma_offset = 0,
 	.rx_dma_offset = 0,
-	.asp_chan_q = EVENTQ_0,
 	.version = MCASP_VERSION_3,
 };
 
 static struct davinci_mcasp_pdata dra7_mcasp_pdata = {
 	.tx_dma_offset = 0x200,
 	.rx_dma_offset = 0x284,
-	.asp_chan_q = EVENTQ_0,
 	.version = MCASP_VERSION_4,
 };
 
@@ -1370,12 +1353,12 @@
 
 static int davinci_mcasp_probe(struct platform_device *pdev)
 {
-	struct davinci_pcm_dma_params *dma_params;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	struct resource *mem, *ioarea, *res, *dat;
 	struct davinci_mcasp_pdata *pdata;
 	struct davinci_mcasp *mcasp;
 	char *irq_name;
+	int *dma;
 	int irq;
 	int ret;
 
@@ -1509,59 +1492,45 @@
 	if (dat)
 		mcasp->dat_port = true;
 
-	dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
 	dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
-	dma_params->asp_chan_q = pdata->asp_chan_q;
-	dma_params->ram_chan_q = pdata->ram_chan_q;
-	dma_params->sram_pool = pdata->sram_pool;
-	dma_params->sram_size = pdata->sram_size_playback;
 	if (dat)
-		dma_params->dma_addr = dat->start;
+		dma_data->addr = dat->start;
 	else
-		dma_params->dma_addr = mem->start + pdata->tx_dma_offset;
+		dma_data->addr = mem->start + pdata->tx_dma_offset;
 
-	/* Unconditional dmaengine stuff */
-	dma_data->addr = dma_params->dma_addr;
-
+	dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (res)
-		dma_params->channel = res->start;
+		*dma = res->start;
 	else
-		dma_params->channel = pdata->tx_dma_channel;
+		*dma = pdata->tx_dma_channel;
 
 	/* dmaengine filter data for DT and non-DT boot */
 	if (pdev->dev.of_node)
 		dma_data->filter_data = "tx";
 	else
-		dma_data->filter_data = &dma_params->channel;
+		dma_data->filter_data = dma;
 
 	/* RX is not valid in DIT mode */
 	if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
-		dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
 		dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
-		dma_params->asp_chan_q = pdata->asp_chan_q;
-		dma_params->ram_chan_q = pdata->ram_chan_q;
-		dma_params->sram_pool = pdata->sram_pool;
-		dma_params->sram_size = pdata->sram_size_capture;
 		if (dat)
-			dma_params->dma_addr = dat->start;
+			dma_data->addr = dat->start;
 		else
-			dma_params->dma_addr = mem->start + pdata->rx_dma_offset;
+			dma_data->addr = mem->start + pdata->rx_dma_offset;
 
-		/* Unconditional dmaengine stuff */
-		dma_data->addr = dma_params->dma_addr;
-
+		dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE];
 		res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 		if (res)
-			dma_params->channel = res->start;
+			*dma = res->start;
 		else
-			dma_params->channel = pdata->rx_dma_channel;
+			*dma = pdata->rx_dma_channel;
 
 		/* dmaengine filter data for DT and non-DT boot */
 		if (pdev->dev.of_node)
 			dma_data->filter_data = "rx";
 		else
-			dma_data->filter_data = &dma_params->channel;
+			dma_data->filter_data = dma;
 	}
 
 	if (mcasp->version < MCASP_VERSION_3) {
@@ -1584,17 +1553,11 @@
 		goto err;
 
 	switch (mcasp->version) {
-#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \
-	(IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
-	 IS_MODULE(CONFIG_SND_DAVINCI_SOC))
-	case MCASP_VERSION_1:
-	case MCASP_VERSION_2:
-		ret = davinci_soc_platform_register(&pdev->dev);
-		break;
-#endif
 #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \
 	(IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
 	 IS_MODULE(CONFIG_SND_EDMA_SOC))
+	case MCASP_VERSION_1:
+	case MCASP_VERSION_2:
 	case MCASP_VERSION_3:
 		ret = edma_pcm_platform_register(&pdev->dev);
 		break;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
deleted file mode 100644
index 7809e9d..0000000
--- a/sound/soc/davinci/davinci-pcm.c
+++ /dev/null
@@ -1,861 +0,0 @@
-/*
- * ALSA PCM interface for the TI DAVINCI processor
- *
- * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
- * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
- * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/kernel.h>
-#include <linux/genalloc.h>
-#include <linux/platform_data/edma.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/dma.h>
-
-#include "davinci-pcm.h"
-
-#ifdef DEBUG
-static void print_buf_info(int slot, char *name)
-{
-	struct edmacc_param p;
-	if (slot < 0)
-		return;
-	edma_read_slot(slot, &p);
-	printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n",
-			name, slot, p.opt, p.src, p.a_b_cnt, p.dst);
-	printk(KERN_DEBUG "    src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n",
-			p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt);
-}
-#else
-static void print_buf_info(int slot, char *name)
-{
-}
-#endif
-
-static struct snd_pcm_hardware pcm_hardware_playback = {
-	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME|
-		 SNDRV_PCM_INFO_BATCH),
-	.buffer_bytes_max = 128 * 1024,
-	.period_bytes_min = 32,
-	.period_bytes_max = 8 * 1024,
-	.periods_min = 16,
-	.periods_max = 255,
-	.fifo_size = 0,
-};
-
-static struct snd_pcm_hardware pcm_hardware_capture = {
-	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_PAUSE |
-		 SNDRV_PCM_INFO_BATCH),
-	.buffer_bytes_max = 128 * 1024,
-	.period_bytes_min = 32,
-	.period_bytes_max = 8 * 1024,
-	.periods_min = 16,
-	.periods_max = 255,
-	.fifo_size = 0,
-};
-
-/*
- * How ping/pong works....
- *
- * Playback:
- * ram_params - copys 2*ping_size from start of SDRAM to iram,
- * 	links to ram_link2
- * ram_link2 - copys rest of SDRAM to iram in ping_size units,
- * 	links to ram_link
- * ram_link - copys entire SDRAM to iram in ping_size uints,
- * 	links to self
- *
- * asp_params - same as asp_link[0]
- * asp_link[0] - copys from lower half of iram to asp port
- * 	links to asp_link[1], triggers iram copy event on completion
- * asp_link[1] - copys from upper half of iram to asp port
- * 	links to asp_link[0], triggers iram copy event on completion
- * 	triggers interrupt only needed to let upper SOC levels update position
- * 	in stream on completion
- *
- * When playback is started:
- * 	ram_params started
- * 	asp_params started
- *
- * Capture:
- * ram_params - same as ram_link,
- * 	links to ram_link
- * ram_link - same as playback
- * 	links to self
- *
- * asp_params - same as playback
- * asp_link[0] - same as playback
- * asp_link[1] - same as playback
- *
- * When capture is started:
- * 	asp_params started
- */
-struct davinci_runtime_data {
-	spinlock_t lock;
-	int period;		/* current DMA period */
-	int asp_channel;	/* Master DMA channel */
-	int asp_link[2];	/* asp parameter link channel, ping/pong */
-	struct davinci_pcm_dma_params *params;	/* DMA params */
-	int ram_channel;
-	int ram_link;
-	int ram_link2;
-	struct edmacc_param asp_params;
-	struct edmacc_param ram_params;
-};
-
-static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream)
-{
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	prtd->period++;
-	if (unlikely(prtd->period >= runtime->periods))
-		prtd->period = 0;
-}
-
-static void davinci_pcm_period_reset(struct snd_pcm_substream *substream)
-{
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-
-	prtd->period = 0;
-}
-/*
- * Not used with ping/pong
- */
-static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
-{
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	unsigned int period_size;
-	unsigned int dma_offset;
-	dma_addr_t dma_pos;
-	dma_addr_t src, dst;
-	unsigned short src_bidx, dst_bidx;
-	unsigned short src_cidx, dst_cidx;
-	unsigned int data_type;
-	unsigned short acnt;
-	unsigned int count;
-	unsigned int fifo_level;
-
-	period_size = snd_pcm_lib_period_bytes(substream);
-	dma_offset = prtd->period * period_size;
-	dma_pos = runtime->dma_addr + dma_offset;
-	fifo_level = prtd->params->fifo_level;
-
-	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
-		"dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos,
-		period_size);
-
-	data_type = prtd->params->data_type;
-	count = period_size / data_type;
-	if (fifo_level)
-		count /= fifo_level;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		src = dma_pos;
-		dst = prtd->params->dma_addr;
-		src_bidx = data_type;
-		dst_bidx = 4;
-		src_cidx = data_type * fifo_level;
-		dst_cidx = 0;
-	} else {
-		src = prtd->params->dma_addr;
-		dst = dma_pos;
-		src_bidx = 0;
-		dst_bidx = data_type;
-		src_cidx = 0;
-		dst_cidx = data_type * fifo_level;
-	}
-
-	acnt = prtd->params->acnt;
-	edma_set_src(prtd->asp_link[0], src, INCR, W8BIT);
-	edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT);
-
-	edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx);
-	edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx);
-
-	if (!fifo_level)
-		edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0,
-							ASYNC);
-	else
-		edma_set_transfer_params(prtd->asp_link[0], acnt,
-						fifo_level,
-						count, fifo_level,
-						ABSYNC);
-}
-
-static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
-{
-	struct snd_pcm_substream *substream = data;
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-
-	print_buf_info(prtd->ram_channel, "i ram_channel");
-	pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status);
-
-	if (unlikely(ch_status != EDMA_DMA_COMPLETE))
-		return;
-
-	if (snd_pcm_running(substream)) {
-		spin_lock(&prtd->lock);
-		if (prtd->ram_channel < 0) {
-			/* No ping/pong must fix up link dma data*/
-			davinci_pcm_enqueue_dma(substream);
-		}
-		davinci_pcm_period_elapsed(substream);
-		spin_unlock(&prtd->lock);
-		snd_pcm_period_elapsed(substream);
-	}
-}
-
-#ifdef CONFIG_GENERIC_ALLOCATOR
-static int allocate_sram(struct snd_pcm_substream *substream,
-		struct gen_pool *sram_pool, unsigned size,
-		struct snd_pcm_hardware *ppcm)
-{
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	struct snd_dma_buffer *iram_dma = NULL;
-	dma_addr_t iram_phys = 0;
-	void *iram_virt = NULL;
-
-	if (buf->private_data || !size)
-		return 0;
-
-	ppcm->period_bytes_max = size;
-	iram_virt = gen_pool_dma_alloc(sram_pool, size, &iram_phys);
-	if (!iram_virt)
-		goto exit1;
-	iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL);
-	if (!iram_dma)
-		goto exit2;
-	iram_dma->area = iram_virt;
-	iram_dma->addr = iram_phys;
-	memset(iram_dma->area, 0, size);
-	iram_dma->bytes = size;
-	buf->private_data = iram_dma;
-	return 0;
-exit2:
-	if (iram_virt)
-		gen_pool_free(sram_pool, (unsigned)iram_virt, size);
-exit1:
-	return -ENOMEM;
-}
-
-static void davinci_free_sram(struct snd_pcm_substream *substream,
-			      struct snd_dma_buffer *iram_dma)
-{
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct gen_pool *sram_pool = prtd->params->sram_pool;
-
-	gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes);
-}
-#else
-static int allocate_sram(struct snd_pcm_substream *substream,
-		struct gen_pool *sram_pool, unsigned size,
-		struct snd_pcm_hardware *ppcm)
-{
-	return 0;
-}
-
-static void davinci_free_sram(struct snd_pcm_substream *substream,
-			      struct snd_dma_buffer *iram_dma)
-{
-}
-#endif
-
-/*
- * Only used with ping/pong.
- * This is called after runtime->dma_addr, period_bytes and data_type are valid
- */
-static int ping_pong_dma_setup(struct snd_pcm_substream *substream)
-{
-	unsigned short ram_src_cidx, ram_dst_cidx;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct davinci_runtime_data *prtd = runtime->private_data;
-	struct snd_dma_buffer *iram_dma =
-		(struct snd_dma_buffer *)substream->dma_buffer.private_data;
-	struct davinci_pcm_dma_params *params = prtd->params;
-	unsigned int data_type = params->data_type;
-	unsigned int acnt = params->acnt;
-	/* divide by 2 for ping/pong */
-	unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1;
-	unsigned int fifo_level = prtd->params->fifo_level;
-	unsigned int count;
-	if ((data_type == 0) || (data_type > 4)) {
-		printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
-		return -EINVAL;
-	}
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		dma_addr_t asp_src_pong = iram_dma->addr + ping_size;
-		ram_src_cidx = ping_size;
-		ram_dst_cidx = -ping_size;
-		edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT);
-
-		edma_set_src_index(prtd->asp_link[0], data_type,
-				data_type * fifo_level);
-		edma_set_src_index(prtd->asp_link[1], data_type,
-				data_type * fifo_level);
-
-		edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT);
-	} else {
-		dma_addr_t asp_dst_pong = iram_dma->addr + ping_size;
-		ram_src_cidx = -ping_size;
-		ram_dst_cidx = ping_size;
-		edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT);
-
-		edma_set_dest_index(prtd->asp_link[0], data_type,
-				data_type * fifo_level);
-		edma_set_dest_index(prtd->asp_link[1], data_type,
-				data_type * fifo_level);
-
-		edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT);
-	}
-
-	if (!fifo_level) {
-		count = ping_size / data_type;
-		edma_set_transfer_params(prtd->asp_link[0], acnt, count,
-				1, 0, ASYNC);
-		edma_set_transfer_params(prtd->asp_link[1], acnt, count,
-				1, 0, ASYNC);
-	} else {
-		count = ping_size / (data_type * fifo_level);
-		edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level,
-				count, fifo_level, ABSYNC);
-		edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level,
-				count, fifo_level, ABSYNC);
-	}
-
-	edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx);
-	edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx);
-	edma_set_transfer_params(prtd->ram_link, ping_size, 2,
-			runtime->periods, 2, ASYNC);
-
-	/* init master params */
-	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
-	edma_read_slot(prtd->ram_link, &prtd->ram_params);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		struct edmacc_param p_ram;
-		/* Copy entire iram buffer before playback started */
-		prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1);
-		/* 0 dst_bidx */
-		prtd->ram_params.src_dst_bidx = (ping_size << 1);
-		/* 0 dst_cidx */
-		prtd->ram_params.src_dst_cidx = (ping_size << 1);
-		prtd->ram_params.ccnt = 1;
-
-		/* Skip 1st period */
-		edma_read_slot(prtd->ram_link, &p_ram);
-		p_ram.src += (ping_size << 1);
-		p_ram.ccnt -= 1;
-		edma_write_slot(prtd->ram_link2, &p_ram);
-		/*
-		 * When 1st started, ram -> iram dma channel will fill the
-		 * entire iram.  Then, whenever a ping/pong asp buffer finishes,
-		 * 1/2 iram will be filled.
-		 */
-		prtd->ram_params.link_bcntrld =
-			EDMA_CHAN_SLOT(prtd->ram_link2) << 5;
-	}
-	return 0;
-}
-
-/* 1 asp tx or rx channel using 2 parameter channels
- * 1 ram to/from iram channel using 1 parameter channel
- *
- * Playback
- * ram copy channel kicks off first,
- * 1st ram copy of entire iram buffer completion kicks off asp channel
- * asp tcc always kicks off ram copy of 1/2 iram buffer
- *
- * Record
- * asp channel starts, tcc kicks off ram copy
- */
-static int request_ping_pong(struct snd_pcm_substream *substream,
-		struct davinci_runtime_data *prtd,
-		struct snd_dma_buffer *iram_dma)
-{
-	dma_addr_t asp_src_ping;
-	dma_addr_t asp_dst_ping;
-	int ret;
-	struct davinci_pcm_dma_params *params = prtd->params;
-
-	/* Request ram master channel */
-	ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY,
-				  davinci_pcm_dma_irq, substream,
-				  prtd->params->ram_chan_q);
-	if (ret < 0)
-		goto exit1;
-
-	/* Request ram link channel */
-	ret = prtd->ram_link = edma_alloc_slot(
-			EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY);
-	if (ret < 0)
-		goto exit2;
-
-	ret = prtd->asp_link[1] = edma_alloc_slot(
-			EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY);
-	if (ret < 0)
-		goto exit3;
-
-	prtd->ram_link2 = -1;
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		ret = prtd->ram_link2 = edma_alloc_slot(
-			EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY);
-		if (ret < 0)
-			goto exit4;
-	}
-	/* circle ping-pong buffers */
-	edma_link(prtd->asp_link[0], prtd->asp_link[1]);
-	edma_link(prtd->asp_link[1], prtd->asp_link[0]);
-	/* circle ram buffers */
-	edma_link(prtd->ram_link, prtd->ram_link);
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		asp_src_ping = iram_dma->addr;
-		asp_dst_ping = params->dma_addr;	/* fifo */
-	} else {
-		asp_src_ping = params->dma_addr;	/* fifo */
-		asp_dst_ping = iram_dma->addr;
-	}
-	/* ping */
-	edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT);
-	edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT);
-	edma_set_src_index(prtd->asp_link[0], 0, 0);
-	edma_set_dest_index(prtd->asp_link[0], 0, 0);
-
-	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
-	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN);
-	prtd->asp_params.opt |= TCCHEN |
-		EDMA_TCC(prtd->ram_channel & 0x3f);
-	edma_write_slot(prtd->asp_link[0], &prtd->asp_params);
-
-	/* pong */
-	edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT);
-	edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT);
-	edma_set_src_index(prtd->asp_link[1], 0, 0);
-	edma_set_dest_index(prtd->asp_link[1], 0, 0);
-
-	edma_read_slot(prtd->asp_link[1], &prtd->asp_params);
-	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f));
-	/* interrupt after every pong completion */
-	prtd->asp_params.opt |= TCINTEN | TCCHEN |
-		EDMA_TCC(prtd->ram_channel & 0x3f);
-	edma_write_slot(prtd->asp_link[1], &prtd->asp_params);
-
-	/* ram */
-	edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT);
-	edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT);
-	pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u,"
-		"for asp:%u %u %u\n", __func__,
-		prtd->ram_channel, prtd->ram_link, prtd->ram_link2,
-		prtd->asp_channel, prtd->asp_link[0],
-		prtd->asp_link[1]);
-	return 0;
-exit4:
-	edma_free_channel(prtd->asp_link[1]);
-	prtd->asp_link[1] = -1;
-exit3:
-	edma_free_channel(prtd->ram_link);
-	prtd->ram_link = -1;
-exit2:
-	edma_free_channel(prtd->ram_channel);
-	prtd->ram_channel = -1;
-exit1:
-	return ret;
-}
-
-static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
-{
-	struct snd_dma_buffer *iram_dma;
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct davinci_pcm_dma_params *params = prtd->params;
-	int ret;
-
-	if (!params)
-		return -ENODEV;
-
-	/* Request asp master DMA channel */
-	ret = prtd->asp_channel = edma_alloc_channel(params->channel,
-			davinci_pcm_dma_irq, substream,
-			prtd->params->asp_chan_q);
-	if (ret < 0)
-		goto exit1;
-
-	/* Request asp link channels */
-	ret = prtd->asp_link[0] = edma_alloc_slot(
-			EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY);
-	if (ret < 0)
-		goto exit2;
-
-	iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data;
-	if (iram_dma) {
-		if (request_ping_pong(substream, prtd, iram_dma) == 0)
-			return 0;
-		printk(KERN_WARNING "%s: dma channel allocation failed,"
-				"not using sram\n", __func__);
-	}
-
-	/* Issue transfer completion IRQ when the channel completes a
-	 * transfer, then always reload from the same slot (by a kind
-	 * of loopback link).  The completion IRQ handler will update
-	 * the reload slot with a new buffer.
-	 *
-	 * REVISIT save p_ram here after setting up everything except
-	 * the buffer and its length (ccnt) ... use it as a template
-	 * so davinci_pcm_enqueue_dma() takes less time in IRQ.
-	 */
-	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
-	prtd->asp_params.opt |= TCINTEN |
-		EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel));
-	prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5;
-	edma_write_slot(prtd->asp_link[0], &prtd->asp_params);
-	return 0;
-exit2:
-	edma_free_channel(prtd->asp_channel);
-	prtd->asp_channel = -1;
-exit1:
-	return ret;
-}
-
-static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	int ret = 0;
-
-	spin_lock(&prtd->lock);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		edma_start(prtd->asp_channel);
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
-		    prtd->ram_channel >= 0) {
-			/* copy 1st iram buffer */
-			edma_start(prtd->ram_channel);
-		}
-		break;
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		edma_resume(prtd->asp_channel);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		edma_pause(prtd->asp_channel);
-		break;
-	default:
-		ret = -EINVAL;
-		break;
-	}
-
-	spin_unlock(&prtd->lock);
-
-	return ret;
-}
-
-static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-
-	davinci_pcm_period_reset(substream);
-	if (prtd->ram_channel >= 0) {
-		int ret = ping_pong_dma_setup(substream);
-		if (ret < 0)
-			return ret;
-
-		edma_write_slot(prtd->ram_channel, &prtd->ram_params);
-		edma_write_slot(prtd->asp_channel, &prtd->asp_params);
-
-		print_buf_info(prtd->ram_channel, "ram_channel");
-		print_buf_info(prtd->ram_link, "ram_link");
-		print_buf_info(prtd->ram_link2, "ram_link2");
-		print_buf_info(prtd->asp_channel, "asp_channel");
-		print_buf_info(prtd->asp_link[0], "asp_link[0]");
-		print_buf_info(prtd->asp_link[1], "asp_link[1]");
-
-		/*
-		 * There is a phase offset of 2 periods between the position
-		 * used by dma setup and the position reported in the pointer
-		 * function.
-		 *
-		 * The phase offset, when not using ping-pong buffers, is due to
-		 * the two consecutive calls to davinci_pcm_enqueue_dma() below.
-		 *
-		 * Whereas here, with ping-pong buffers, the phase is due to
-		 * there being an entire buffer transfer complete before the
-		 * first dma completion event triggers davinci_pcm_dma_irq().
-		 */
-		davinci_pcm_period_elapsed(substream);
-		davinci_pcm_period_elapsed(substream);
-
-		return 0;
-	}
-	davinci_pcm_enqueue_dma(substream);
-	davinci_pcm_period_elapsed(substream);
-
-	/* Copy self-linked parameter RAM entry into master channel */
-	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
-	edma_write_slot(prtd->asp_channel, &prtd->asp_params);
-	davinci_pcm_enqueue_dma(substream);
-	davinci_pcm_period_elapsed(substream);
-
-	return 0;
-}
-
-static snd_pcm_uframes_t
-davinci_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct davinci_runtime_data *prtd = runtime->private_data;
-	unsigned int offset;
-	int asp_count;
-	unsigned int period_size = snd_pcm_lib_period_bytes(substream);
-
-	/*
-	 * There is a phase offset of 2 periods between the position used by dma
-	 * setup and the position reported in the pointer function. Either +2 in
-	 * the dma setup or -2 here in the pointer function (with wrapping,
-	 * both) accounts for this offset -- choose the latter since it makes
-	 * the first-time setup clearer.
-	 */
-	spin_lock(&prtd->lock);
-	asp_count = prtd->period - 2;
-	spin_unlock(&prtd->lock);
-
-	if (asp_count < 0)
-		asp_count += runtime->periods;
-	asp_count *= period_size;
-
-	offset = bytes_to_frames(runtime, asp_count);
-	if (offset >= runtime->buffer_size)
-		offset = 0;
-
-	return offset;
-}
-
-static int davinci_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct davinci_runtime_data *prtd;
-	struct snd_pcm_hardware *ppcm;
-	int ret = 0;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct davinci_pcm_dma_params *pa;
-	struct davinci_pcm_dma_params *params;
-
-	pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
-	if (!pa)
-		return -ENODEV;
-	params = &pa[substream->stream];
-
-	ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-			&pcm_hardware_playback : &pcm_hardware_capture;
-	allocate_sram(substream, params->sram_pool, params->sram_size, ppcm);
-	snd_soc_set_runtime_hwparams(substream, ppcm);
-	/* ensure that buffer size is a multiple of period size */
-	ret = snd_pcm_hw_constraint_integer(runtime,
-						SNDRV_PCM_HW_PARAM_PERIODS);
-	if (ret < 0)
-		return ret;
-
-	prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL);
-	if (prtd == NULL)
-		return -ENOMEM;
-
-	spin_lock_init(&prtd->lock);
-	prtd->params = params;
-	prtd->asp_channel = -1;
-	prtd->asp_link[0] = prtd->asp_link[1] = -1;
-	prtd->ram_channel = -1;
-	prtd->ram_link = -1;
-	prtd->ram_link2 = -1;
-
-	runtime->private_data = prtd;
-
-	ret = davinci_pcm_dma_request(substream);
-	if (ret) {
-		printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n");
-		kfree(prtd);
-	}
-
-	return ret;
-}
-
-static int davinci_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct davinci_runtime_data *prtd = runtime->private_data;
-
-	if (prtd->ram_channel >= 0)
-		edma_stop(prtd->ram_channel);
-	if (prtd->asp_channel >= 0)
-		edma_stop(prtd->asp_channel);
-	if (prtd->asp_link[0] >= 0)
-		edma_unlink(prtd->asp_link[0]);
-	if (prtd->asp_link[1] >= 0)
-		edma_unlink(prtd->asp_link[1]);
-	if (prtd->ram_link >= 0)
-		edma_unlink(prtd->ram_link);
-
-	if (prtd->asp_link[0] >= 0)
-		edma_free_slot(prtd->asp_link[0]);
-	if (prtd->asp_link[1] >= 0)
-		edma_free_slot(prtd->asp_link[1]);
-	if (prtd->asp_channel >= 0)
-		edma_free_channel(prtd->asp_channel);
-	if (prtd->ram_link >= 0)
-		edma_free_slot(prtd->ram_link);
-	if (prtd->ram_link2 >= 0)
-		edma_free_slot(prtd->ram_link2);
-	if (prtd->ram_channel >= 0)
-		edma_free_channel(prtd->ram_channel);
-
-	kfree(prtd);
-
-	return 0;
-}
-
-static int davinci_pcm_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *hw_params)
-{
-	return snd_pcm_lib_malloc_pages(substream,
-					params_buffer_bytes(hw_params));
-}
-
-static int davinci_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	return snd_pcm_lib_free_pages(substream);
-}
-
-static int davinci_pcm_mmap(struct snd_pcm_substream *substream,
-			    struct vm_area_struct *vma)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-
-	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
-				     runtime->dma_area,
-				     runtime->dma_addr,
-				     runtime->dma_bytes);
-}
-
-static struct snd_pcm_ops davinci_pcm_ops = {
-	.open = 	davinci_pcm_open,
-	.close = 	davinci_pcm_close,
-	.ioctl = 	snd_pcm_lib_ioctl,
-	.hw_params = 	davinci_pcm_hw_params,
-	.hw_free = 	davinci_pcm_hw_free,
-	.prepare = 	davinci_pcm_prepare,
-	.trigger = 	davinci_pcm_trigger,
-	.pointer = 	davinci_pcm_pointer,
-	.mmap = 	davinci_pcm_mmap,
-};
-
-static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
-		size_t size)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *buf = &substream->dma_buffer;
-
-	buf->dev.type = SNDRV_DMA_TYPE_DEV;
-	buf->dev.dev = pcm->card->dev;
-	buf->private_data = NULL;
-	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
-					   &buf->addr, GFP_KERNEL);
-
-	pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, "
-		"size=%d\n", (void *) buf->area, (void *) buf->addr, size);
-
-	if (!buf->area)
-		return -ENOMEM;
-
-	buf->bytes = size;
-	return 0;
-}
-
-static void davinci_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_pcm_substream *substream;
-	struct snd_dma_buffer *buf;
-	int stream;
-
-	for (stream = 0; stream < 2; stream++) {
-		struct snd_dma_buffer *iram_dma;
-		substream = pcm->streams[stream].substream;
-		if (!substream)
-			continue;
-
-		buf = &substream->dma_buffer;
-		if (!buf->area)
-			continue;
-
-		dma_free_writecombine(pcm->card->dev, buf->bytes,
-				      buf->area, buf->addr);
-		buf->area = NULL;
-		iram_dma = buf->private_data;
-		if (iram_dma) {
-			davinci_free_sram(substream, iram_dma);
-			kfree(iram_dma);
-		}
-	}
-}
-
-static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_card *card = rtd->card->snd_card;
-	struct snd_pcm *pcm = rtd->pcm;
-	int ret;
-
-	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
-	if (ret)
-		return ret;
-
-	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
-		ret = davinci_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK,
-			pcm_hardware_playback.buffer_bytes_max);
-		if (ret)
-			return ret;
-	}
-
-	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
-		ret = davinci_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE,
-			pcm_hardware_capture.buffer_bytes_max);
-		if (ret)
-			return ret;
-	}
-
-	return 0;
-}
-
-static struct snd_soc_platform_driver davinci_soc_platform = {
-	.ops =		&davinci_pcm_ops,
-	.pcm_new = 	davinci_pcm_new,
-	.pcm_free = 	davinci_pcm_free,
-};
-
-int davinci_soc_platform_register(struct device *dev)
-{
-	return devm_snd_soc_register_platform(dev, &davinci_soc_platform);
-}
-EXPORT_SYMBOL_GPL(davinci_soc_platform_register);
-
-MODULE_AUTHOR("Vladimir Barinov");
-MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
deleted file mode 100644
index 0fe2346..0000000
--- a/sound/soc/davinci/davinci-pcm.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * ALSA PCM interface for the TI DAVINCI processor
- *
- * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
- * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _DAVINCI_PCM_H
-#define _DAVINCI_PCM_H
-
-#include <linux/genalloc.h>
-#include <linux/platform_data/davinci_asp.h>
-#include <linux/platform_data/edma.h>
-
-struct davinci_pcm_dma_params {
-	int channel;			/* sync dma channel ID */
-	unsigned short acnt;
-	dma_addr_t dma_addr;		/* device physical address for DMA */
-	unsigned sram_size;
-	struct gen_pool *sram_pool;	/* SRAM gen_pool for ping pong */
-	enum dma_event_q asp_chan_q;	/* event queue number for ASP channel */
-	enum dma_event_q ram_chan_q;	/* event queue number for RAM channel */
-	unsigned char data_type;	/* xfer data type */
-	unsigned char convert_mono_stereo;
-	unsigned int fifo_level;
-};
-
-#if IS_ENABLED(CONFIG_SND_DAVINCI_SOC)
-int davinci_soc_platform_register(struct device *dev);
-#else
-static inline int davinci_soc_platform_register(struct device *dev)
-{
-	return 0;
-}
-#endif /* CONFIG_SND_DAVINCI_SOC */
-
-#endif
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 5bee0427..fabd05f 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -33,8 +33,9 @@
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
-#include "davinci-pcm.h"
+#include "edma-pcm.h"
 #include "davinci-i2s.h"
 
 #define MOD_REG_BIT(val, mask, set) do { \
@@ -47,7 +48,8 @@
 
 struct davinci_vcif_dev {
 	struct davinci_vc *davinci_vc;
-	struct davinci_pcm_dma_params	dma_params[2];
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+	int dma_request[2];
 };
 
 static void davinci_vcif_start(struct snd_pcm_substream *substream)
@@ -93,8 +95,6 @@
 {
 	struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai);
 	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
-	struct davinci_pcm_dma_params *dma_params =
-			&davinci_vcif_dev->dma_params[substream->stream];
 	u32 w;
 
 	/* Restart the codec before setup */
@@ -113,16 +113,12 @@
 	/* Determine xfer data type */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U8:
-		dma_params->data_type = 0;
-
 		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
 			    DAVINCI_VC_CTRL_RD_UNSIGNED |
 			    DAVINCI_VC_CTRL_WD_BITS_8 |
 			    DAVINCI_VC_CTRL_WD_UNSIGNED, 1);
 		break;
 	case SNDRV_PCM_FORMAT_S8:
-		dma_params->data_type = 1;
-
 		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
 			    DAVINCI_VC_CTRL_WD_BITS_8, 1);
 
@@ -130,8 +126,6 @@
 			    DAVINCI_VC_CTRL_WD_UNSIGNED, 0);
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
-		dma_params->data_type = 2;
-
 		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
 			    DAVINCI_VC_CTRL_RD_UNSIGNED |
 			    DAVINCI_VC_CTRL_WD_BITS_8 |
@@ -142,8 +136,6 @@
 		return -EINVAL;
 	}
 
-	dma_params->acnt  = dma_params->data_type;
-
 	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
 
 	return 0;
@@ -172,24 +164,25 @@
 	return ret;
 }
 
-static int davinci_vcif_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
-
-	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
-	return 0;
-}
-
 #define DAVINCI_VCIF_RATES	SNDRV_PCM_RATE_8000_48000
 
 static const struct snd_soc_dai_ops davinci_vcif_dai_ops = {
-	.startup	= davinci_vcif_startup,
 	.trigger	= davinci_vcif_trigger,
 	.hw_params	= davinci_vcif_hw_params,
 };
 
+static int davinci_vcif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+
+	return 0;
+}
+
 static struct snd_soc_dai_driver davinci_vcif_dai = {
+	.probe = davinci_vcif_dai_probe,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
@@ -225,16 +218,16 @@
 
 	/* DMA tx params */
 	davinci_vcif_dev->davinci_vc = davinci_vc;
-	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel =
-					davinci_vc->davinci_vcif.dma_tx_channel;
-	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
-					davinci_vc->davinci_vcif.dma_tx_addr;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data =
+				&davinci_vc->davinci_vcif.dma_tx_channel;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
+				davinci_vc->davinci_vcif.dma_tx_addr;
 
 	/* DMA rx params */
-	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel =
-					davinci_vc->davinci_vcif.dma_rx_channel;
-	davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
-					davinci_vc->davinci_vcif.dma_rx_addr;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data =
+				&davinci_vc->davinci_vcif.dma_rx_channel;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
+				davinci_vc->davinci_vcif.dma_rx_addr;
 
 	dev_set_drvdata(&pdev->dev, davinci_vcif_dev);
 
@@ -245,7 +238,7 @@
 		return ret;
 	}
 
-	ret = davinci_soc_platform_register(&pdev->dev);
+	ret = edma_pcm_platform_register(&pdev->dev);
 	if (ret) {
 		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
 		snd_soc_unregister_component(&pdev->dev);
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 3f6959c..de43887 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -512,6 +512,12 @@
 	memcpy(priv->dai_link, fsl_asoc_card_dai,
 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
 
+	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+		goto asrc_fail;
+	}
+
 	/* Normal DAI Link */
 	priv->dai_link[0].cpu_of_node = cpu_np;
 	priv->dai_link[0].codec_of_node = codec_np;
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index f8cf10e..20e7400 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -53,9 +53,9 @@
 
 	/* Headphone jack detection */
 	if (gpio_is_valid(data->jack_gpio)) {
-		ret = snd_soc_jack_new(rtd->codec, "Headphone",
-				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
-				       &headset_jack);
+		ret = snd_soc_card_jack_new(rtd->card, "Headphone",
+					    SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+					    &headset_jack, NULL, 0);
 		if (ret)
 			return ret;
 
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index a958937a..0653aa8 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -205,16 +205,14 @@
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	/* Headphone jack detection */
-	snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
-	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
-			      hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
+			      &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
 	wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
 
 	/* Microphone jack detection */
-	snd_soc_jack_new(codec, "Microphone",
-			 SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
-	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
-			      mic_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Microphone",
+			      SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
+			      mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
 	wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
 			       SND_JACK_BTN_0);
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index fb550b5..c49a408 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -176,11 +176,11 @@
 		return ret;
 
 	if (gpio_is_valid(priv->gpio_hp_det)) {
-		snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE,
-				 &simple_card_hp_jack);
-		snd_soc_jack_add_pins(&simple_card_hp_jack,
-				      ARRAY_SIZE(simple_card_hp_jack_pins),
-				      simple_card_hp_jack_pins);
+		snd_soc_card_jack_new(rtd->card, "Headphones",
+				      SND_JACK_HEADPHONE,
+				      &simple_card_hp_jack,
+				      simple_card_hp_jack_pins,
+				      ARRAY_SIZE(simple_card_hp_jack_pins));
 
 		simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det;
 		simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert;
@@ -189,11 +189,11 @@
 	}
 
 	if (gpio_is_valid(priv->gpio_mic_det)) {
-		snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE,
-				 &simple_card_mic_jack);
-		snd_soc_jack_add_pins(&simple_card_mic_jack,
-				      ARRAY_SIZE(simple_card_mic_jack_pins),
-				      simple_card_mic_jack_pins);
+		snd_soc_card_jack_new(rtd->card, "Mic Jack",
+				      SND_JACK_MICROPHONE,
+				      &simple_card_mic_jack,
+				      simple_card_mic_jack_pins,
+				      ARRAY_SIZE(simple_card_mic_jack_pins));
 		simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det;
 		simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert;
 		snd_soc_jack_add_gpios(&simple_card_mic_jack, 1,
diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c
index 9cf7d01..fc55420 100644
--- a/sound/soc/intel/broadwell.c
+++ b/sound/soc/intel/broadwell.c
@@ -80,15 +80,9 @@
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	int ret = 0;
-	ret = snd_soc_jack_new(codec, "Headset",
-		SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset);
-
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(&broadwell_headset,
-		ARRAY_SIZE(broadwell_headset_pins),
-		broadwell_headset_pins);
+	ret = snd_soc_card_jack_new(rtd->card, "Headset",
+		SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset,
+		broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins));
 	if (ret)
 		return ret;
 
@@ -110,9 +104,7 @@
 	channels->min = channels->max = 2;
 
 	/* set SSP0 to 16 bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c
index 9832afe..d8b1f03 100644
--- a/sound/soc/intel/byt-max98090.c
+++ b/sound/soc/intel/byt-max98090.c
@@ -84,7 +84,6 @@
 static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
 {
 	int ret;
-	struct snd_soc_codec *codec = runtime->codec;
 	struct snd_soc_card *card = runtime->card;
 	struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card);
 	struct snd_soc_jack *jack = &drv->jack;
@@ -100,13 +99,9 @@
 	}
 
 	/* Enable jack detection */
-	ret = snd_soc_jack_new(codec, "Headset",
-			       SND_JACK_LINEOUT | SND_JACK_HEADSET, jack);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins),
-				    hs_jack_pins);
+	ret = snd_soc_card_jack_new(runtime->card, "Headset",
+				    SND_JACK_LINEOUT | SND_JACK_HEADSET, jack,
+				    hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c
index 5930862..3b262d0 100644
--- a/sound/soc/intel/bytcr_dpcm_rt5640.c
+++ b/sound/soc/intel/bytcr_dpcm_rt5640.c
@@ -113,9 +113,7 @@
 	channels->min = channels->max = 2;
 
 	/* set SSP2 to 24-bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S24_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c
index bd29617..0122279 100644
--- a/sound/soc/intel/cht_bsw_rt5645.c
+++ b/sound/soc/intel/cht_bsw_rt5645.c
@@ -169,17 +169,17 @@
 		return ret;
 	}
 
-	ret = snd_soc_jack_new(codec, "Headphone Jack",
-				SND_JACK_HEADPHONE,
-				&ctx->hp_jack);
+	ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &ctx->hp_jack,
+				    NULL, 0);
 	if (ret) {
 		dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
 		return ret;
 	}
 
-	ret = snd_soc_jack_new(codec, "Mic Jack",
-				SND_JACK_MICROPHONE,
-				&ctx->mic_jack);
+	ret = snd_soc_card_jack_new(runtime->card, "Mic Jack",
+				    SND_JACK_MICROPHONE, &ctx->mic_jack,
+				    NULL, 0);
 	if (ret) {
 		dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
 		return ret;
@@ -203,9 +203,7 @@
 	channels->min = channels->max = 2;
 
 	/* set SSP2 to 24-bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S24_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c
index ff01662..bc8dcac 100644
--- a/sound/soc/intel/cht_bsw_rt5672.c
+++ b/sound/soc/intel/cht_bsw_rt5672.c
@@ -178,9 +178,7 @@
 	channels->min = channels->max = 2;
 
 	/* set SSP2 to 24-bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S24_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
 	return 0;
 }
 
@@ -217,7 +215,7 @@
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.codec_name = "snd-soc-dummy",
 		.platform_name = "sst-mfld-platform",
-		.ignore_suspend = 1,
+		.nonatomic = true,
 		.dynamic = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
@@ -240,13 +238,13 @@
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
+		.nonatomic = true,
 		.codec_dai_name = "rt5670-aif1",
 		.codec_name = "i2c-10EC5670:00",
 		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
 					| SND_SOC_DAIFMT_CBS_CFS,
 		.init = cht_codec_init,
 		.be_hw_params_fixup = cht_codec_fixup,
-		.ignore_suspend = 1,
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.ops = &cht_be_ssp2_ops,
@@ -285,7 +283,6 @@
 static struct platform_driver snd_cht_mc_driver = {
 	.driver = {
 		.name = "cht-bsw-rt5672",
-		.pm = &snd_soc_pm_ops,
 	},
 	.probe = snd_cht_mc_probe,
 };
diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c
index 35edf51..00fddd3 100644
--- a/sound/soc/intel/haswell.c
+++ b/sound/soc/intel/haswell.c
@@ -56,9 +56,7 @@
 	channels->min = channels->max = 2;
 
 	/* set SSP0 to 16 bit */
-	snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
+	params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
 	return 0;
 }
 
diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c
index 90b7a57..49c09a0 100644
--- a/sound/soc/intel/mfld_machine.c
+++ b/sound/soc/intel/mfld_machine.c
@@ -228,10 +228,13 @@
 {
 	struct mfld_jack_data jack_data;
 
+	if (!mfld_codec)
+		return;
+
 	jack_data.mfld_jack = &mfld_jack;
 	jack_data.intr_id = intr_status;
 
-	sn95031_jack_detection(&jack_data);
+	sn95031_jack_detection(mfld_codec, &jack_data);
 	/* TODO: add american headset detection post gpiolib support */
 }
 
@@ -240,8 +243,6 @@
 	struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
 	int ret_val;
 
-	mfld_codec = runtime->codec;
-
 	/* default is earpiece pin, userspace sets it explcitly */
 	snd_soc_dapm_disable_pin(dapm, "Headphones");
 	/* default is lineout NC, userspace sets it explcitly */
@@ -254,20 +255,15 @@
 	snd_soc_dapm_disable_pin(dapm, "LINEINR");
 
 	/* Headset and button jack detection */
-	ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack",
-			SND_JACK_HEADSET | SND_JACK_BTN_0 |
-			SND_JACK_BTN_1, &mfld_jack);
+	ret_val = snd_soc_card_jack_new(runtime->card,
+			"Intel(R) MID Audio Jack", SND_JACK_HEADSET |
+			SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack,
+			mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins));
 	if (ret_val) {
 		pr_err("jack creation failed\n");
 		return ret_val;
 	}
 
-	ret_val = snd_soc_jack_add_pins(&mfld_jack,
-			ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
-	if (ret_val) {
-		pr_err("adding jack pins failed\n");
-		return ret_val;
-	}
 	ret_val = snd_soc_jack_add_zones(&mfld_jack,
 			ARRAY_SIZE(mfld_zones), mfld_zones);
 	if (ret_val) {
@@ -275,6 +271,8 @@
 		return ret_val;
 	}
 
+	mfld_codec = runtime->codec;
+
 	/* we want to check if anything is inserted at boot,
 	 * so send a fake event to codec and it will read adc
 	 * to find if anything is there or not */
@@ -359,8 +357,6 @@
 {
 	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
 
-	if (mfld_jack.codec == NULL)
-		return IRQ_HANDLED;
 	mfld_jack_check(mc_drv_ctx->interrupt_status);
 
 	return IRQ_HANDLED;
diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c
index 7523cbe..2fbaf2c 100644
--- a/sound/soc/intel/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/sst-mfld-platform-pcm.c
@@ -594,11 +594,13 @@
 		ret_val = stream->ops->stream_drop(sst->dev, str_id);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 		dev_dbg(rtd->dev, "sst: in pause\n");
 		status = SST_PLATFORM_PAUSED;
 		ret_val = stream->ops->stream_pause(sst->dev, str_id);
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
 		dev_dbg(rtd->dev, "sst: in pause release\n");
 		status = SST_PLATFORM_RUNNING;
 		ret_val = stream->ops->stream_pause_release(sst->dev, str_id);
@@ -665,6 +667,9 @@
 
 static int sst_soc_probe(struct snd_soc_platform *platform)
 {
+	struct sst_data *drv = dev_get_drvdata(platform->dev);
+
+	drv->soc_card = platform->component.card;
 	return sst_dsp_init_v2_dpcm(platform);
 }
 
@@ -727,9 +732,64 @@
 	return 0;
 }
 
+#ifdef CONFIG_PM_SLEEP
+
+static int sst_soc_prepare(struct device *dev)
+{
+	struct sst_data *drv = dev_get_drvdata(dev);
+	int i;
+
+	/* suspend all pcms first */
+	snd_soc_suspend(drv->soc_card->dev);
+	snd_soc_poweroff(drv->soc_card->dev);
+
+	/* set the SSPs to idle */
+	for (i = 0; i < drv->soc_card->num_rtd; i++) {
+		struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai;
+
+		if (dai->active) {
+			send_ssp_cmd(dai, dai->name, 0);
+			sst_handle_vb_timer(dai, false);
+		}
+	}
+
+	return 0;
+}
+
+static void sst_soc_complete(struct device *dev)
+{
+	struct sst_data *drv = dev_get_drvdata(dev);
+	int i;
+
+	/* restart SSPs */
+	for (i = 0; i < drv->soc_card->num_rtd; i++) {
+		struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai;
+
+		if (dai->active) {
+			sst_handle_vb_timer(dai, true);
+			send_ssp_cmd(dai, dai->name, 1);
+		}
+	}
+	snd_soc_resume(drv->soc_card->dev);
+}
+
+#else
+
+#define sst_soc_prepare NULL
+#define sst_soc_complete NULL
+
+#endif
+
+
+static const struct dev_pm_ops sst_platform_pm = {
+	.prepare	= sst_soc_prepare,
+	.complete	= sst_soc_complete,
+};
+
 static struct platform_driver sst_platform_driver = {
 	.driver		= {
 		.name		= "sst-mfld-platform",
+		.pm             = &sst_platform_pm,
 	},
 	.probe		= sst_platform_probe,
 	.remove		= sst_platform_remove,
diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h
index 79c8d12..9094314 100644
--- a/sound/soc/intel/sst-mfld-platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -174,6 +174,7 @@
 	struct sst_platform_data *pdata;
 	struct snd_sst_bytes_v2 *byte_stream;
 	struct mutex lock;
+	struct snd_soc_card *soc_card;
 };
 int sst_register_dsp(struct sst_device *sst);
 int sst_unregister_dsp(struct sst_device *sst);
diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 11c5786..1a7eeec 100644
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -423,23 +423,135 @@
 	return ret;
 }
 
-static int intel_sst_runtime_resume(struct device *dev)
+static int intel_sst_suspend(struct device *dev)
 {
-	int ret = 0;
 	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	struct sst_fw_save *fw_save;
+	int i, ret = 0;
 
-	if (ctx->sst_state == SST_RESET) {
-		ret = sst_load_fw(ctx);
-		if (ret) {
-			dev_err(dev, "FW download fail %d\n", ret);
-			sst_set_fw_state_locked(ctx, SST_RESET);
+	/* check first if we are already in SW reset */
+	if (ctx->sst_state == SST_RESET)
+		return 0;
+
+	/*
+	 * check if any stream is active and running
+	 * they should already by suspend by soc_suspend
+	 */
+	for (i = 1; i <= ctx->info.max_streams; i++) {
+		struct stream_info *stream = &ctx->streams[i];
+
+		if (stream->status == STREAM_RUNNING) {
+			dev_err(dev, "stream %d is running, cant susupend, abort\n", i);
+			return -EBUSY;
 		}
 	}
+	synchronize_irq(ctx->irq_num);
+	flush_workqueue(ctx->post_msg_wq);
+
+	/* Move the SST state to Reset */
+	sst_set_fw_state_locked(ctx, SST_RESET);
+
+	/* tell DSP we are suspending */
+	if (ctx->ops->save_dsp_context(ctx))
+		return -EBUSY;
+
+	/* save the memories */
+	fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL);
+	if (!fw_save)
+		return -ENOMEM;
+	fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL);
+	if (!fw_save->iram) {
+		ret = -ENOMEM;
+		goto iram;
+	}
+	fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL);
+	if (!fw_save->dram) {
+		ret = -ENOMEM;
+		goto dram;
+	}
+	fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL);
+	if (!fw_save->sram) {
+		ret = -ENOMEM;
+		goto sram;
+	}
+
+	fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL);
+	if (!fw_save->ddr) {
+		ret = -ENOMEM;
+		goto ddr;
+	}
+
+	memcpy32_fromio(fw_save->iram, ctx->iram, ctx->iram_end - ctx->iram_base);
+	memcpy32_fromio(fw_save->dram, ctx->dram, ctx->dram_end - ctx->dram_base);
+	memcpy32_fromio(fw_save->sram, ctx->mailbox, SST_MAILBOX_SIZE);
+	memcpy32_fromio(fw_save->ddr, ctx->ddr, ctx->ddr_end - ctx->ddr_base);
+
+	ctx->fw_save = fw_save;
+	ctx->ops->reset(ctx);
+	return 0;
+ddr:
+	kfree(fw_save->sram);
+sram:
+	kfree(fw_save->dram);
+dram:
+	kfree(fw_save->iram);
+iram:
+	kfree(fw_save);
+	return ret;
+}
+
+static int intel_sst_resume(struct device *dev)
+{
+	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	struct sst_fw_save *fw_save = ctx->fw_save;
+	int ret = 0;
+	struct sst_block *block;
+
+	if (!fw_save)
+		return 0;
+
+	sst_set_fw_state_locked(ctx, SST_FW_LOADING);
+
+	/* we have to restore the memory saved */
+	ctx->ops->reset(ctx);
+
+	ctx->fw_save = NULL;
+
+	memcpy32_toio(ctx->iram, fw_save->iram, ctx->iram_end - ctx->iram_base);
+	memcpy32_toio(ctx->dram, fw_save->dram, ctx->dram_end - ctx->dram_base);
+	memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE);
+	memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base);
+
+	kfree(fw_save->sram);
+	kfree(fw_save->dram);
+	kfree(fw_save->iram);
+	kfree(fw_save->ddr);
+	kfree(fw_save);
+
+	block = sst_create_block(ctx, 0, FW_DWNL_ID);
+	if (block == NULL)
+		return -ENOMEM;
+
+
+	/* start and wait for ack */
+	ctx->ops->start(ctx);
+	ret = sst_wait_timeout(ctx, block);
+	if (ret) {
+		dev_err(ctx->dev, "fw download failed %d\n", ret);
+		/* FW download failed due to timeout */
+		ret = -EBUSY;
+
+	} else {
+		sst_set_fw_state_locked(ctx, SST_FW_RUNNING);
+	}
+
+	sst_free_block(ctx, block);
 	return ret;
 }
 
 const struct dev_pm_ops intel_sst_pm = {
+	.suspend = intel_sst_suspend,
+	.resume = intel_sst_resume,
 	.runtime_suspend = intel_sst_runtime_suspend,
-	.runtime_resume = intel_sst_runtime_resume,
 };
 EXPORT_SYMBOL_GPL(intel_sst_pm);
diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h
index 562bc48..3f49386 100644
--- a/sound/soc/intel/sst/sst.h
+++ b/sound/soc/intel/sst/sst.h
@@ -337,6 +337,13 @@
 	u64 csr2;
 };
 
+struct sst_fw_save {
+	void *iram;
+	void *dram;
+	void *sram;
+	void *ddr;
+};
+
 /**
  * struct intel_sst_drv - driver ops
  *
@@ -428,6 +435,8 @@
 	 * persistent till worker thread gets called
 	 */
 	char firmware_name[FW_NAME_SIZE];
+
+	struct sst_fw_save	*fw_save;
 };
 
 /* misc definitions */
@@ -544,4 +553,7 @@
 int sst_context_init(struct intel_sst_drv *ctx);
 void sst_context_cleanup(struct intel_sst_drv *ctx);
 void sst_configure_runtime_pm(struct intel_sst_drv *ctx);
+void memcpy32_toio(void __iomem *dst, const void *src, int count);
+void memcpy32_fromio(void *dst, const void __iomem *src, int count);
+
 #endif
diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c
index 5f75ef3..f0e4b99b 100644
--- a/sound/soc/intel/sst/sst_drv_interface.c
+++ b/sound/soc/intel/sst/sst_drv_interface.c
@@ -138,12 +138,36 @@
 static int sst_power_control(struct device *dev, bool state)
 {
 	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+	int ret = 0;
+	int usage_count = 0;
 
-	dev_dbg(ctx->dev, "state:%d", state);
-	if (state == true)
-		return pm_runtime_get_sync(dev);
-	else
+#ifdef CONFIG_PM
+	usage_count = atomic_read(&dev->power.usage_count);
+#else
+	usage_count = 1;
+#endif
+
+	if (state == true) {
+		ret = pm_runtime_get_sync(dev);
+
+		dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count);
+		if (ret < 0) {
+			dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret);
+			return ret;
+		}
+		if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) {
+			ret = sst_load_fw(ctx);
+			if (ret) {
+				dev_err(dev, "FW download fail %d\n", ret);
+				sst_set_fw_state_locked(ctx, SST_RESET);
+				ret = sst_pm_runtime_put(ctx);
+			}
+		}
+	} else {
+		dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count);
 		return sst_pm_runtime_put(ctx);
+	}
+	return ret;
 }
 
 /*
@@ -572,6 +596,35 @@
 	return sst_drop_stream(ctx, str_id);
 }
 
+static int sst_stream_pause(struct device *dev, int str_id)
+{
+	struct stream_info *str_info;
+	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+
+	if (ctx->sst_state != SST_FW_RUNNING)
+		return 0;
+
+	str_info = get_stream_info(ctx, str_id);
+	if (!str_info)
+		return -EINVAL;
+
+	return sst_pause_stream(ctx, str_id);
+}
+
+static int sst_stream_resume(struct device *dev, int str_id)
+{
+	struct stream_info *str_info;
+	struct intel_sst_drv *ctx = dev_get_drvdata(dev);
+
+	if (ctx->sst_state != SST_FW_RUNNING)
+		return 0;
+
+	str_info = get_stream_info(ctx, str_id);
+	if (!str_info)
+		return -EINVAL;
+	return sst_resume_stream(ctx, str_id);
+}
+
 static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info)
 {
 	int str_id = 0;
@@ -633,6 +686,8 @@
 	.stream_init = sst_stream_init,
 	.stream_start = sst_stream_start,
 	.stream_drop = sst_stream_drop,
+	.stream_pause = sst_stream_pause,
+	.stream_pause_release = sst_stream_resume,
 	.stream_read_tstamp = sst_read_timestamp,
 	.send_byte_stream = sst_send_byte_stream,
 	.close = sst_close_pcm_stream,
diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c
index 7888cd7..e88907a 100644
--- a/sound/soc/intel/sst/sst_loader.c
+++ b/sound/soc/intel/sst/sst_loader.c
@@ -39,7 +39,15 @@
 #include "sst.h"
 #include "../sst-dsp.h"
 
-static inline void memcpy32_toio(void __iomem *dst, const void *src, int count)
+void memcpy32_toio(void __iomem *dst, const void *src, int count)
+{
+	/* __iowrite32_copy uses 32-bit count values so divide by 4 for
+	 * right count in words
+	 */
+	__iowrite32_copy(dst, src, count/4);
+}
+
+void memcpy32_fromio(void *dst, const void __iomem *src, int count)
 {
 	/* __iowrite32_copy uses 32-bit count values so divide by 4 for
 	 * right count in words
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index a2cd348..e7c78b0 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -100,17 +100,19 @@
 
 config SND_OMAP_SOC_OMAP_ABE_TWL6040
 	tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
-	depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST)
+	depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST)
 	select SND_OMAP_SOC_DMIC
 	select SND_OMAP_SOC_MCPDM
 	select SND_SOC_TWL6040
 	select SND_SOC_DMIC
+	select COMMON_CLK_PALMAS if SOC_OMAP5
 	help
 	  Say Y if you want to add support for SoC audio on OMAP boards using
 	  ABE and twl6040 codec. This driver currently supports:
 	  - SDP4430/Blaze boards
 	  - PandaBoard (4430)
 	  - PandaBoardES (4460)
+	  - omap5-uevm (5432)
 
 config SND_OMAP_SOC_OMAP3_PANDORA
 	tristate "SoC Audio support for OMAP3 Pandora"
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 7066130..16cc95f 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -479,8 +479,8 @@
 
 	/* Add hook switch - can be used to control the codec from userspace
 	 * even if line discipline fails */
-	ret = snd_soc_jack_new(rtd->codec, "hook_switch",
-				SND_JACK_HEADSET, &ams_delta_hook_switch);
+	ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET,
+				    &ams_delta_hook_switch, NULL, 0);
 	if (ret)
 		dev_warn(card->dev,
 				"Failed to allocate resources for hook switch, "
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index b9c65f1..0843a68 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -182,17 +182,17 @@
 
 	/* Headset jack detection only if it is supported */
 	if (priv->jack_detection) {
-		ret = snd_soc_jack_new(codec, "Headset Jack",
-					SND_JACK_HEADSET, &hs_jack);
+		ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+					    SND_JACK_HEADSET, &hs_jack,
+					    hs_jack_pins,
+					    ARRAY_SIZE(hs_jack_pins));
 		if (ret)
 			return ret;
 
-		ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-					hs_jack_pins);
 		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
 	}
 
-	return ret;
+	return 0;
 }
 
 static const struct snd_soc_dapm_route dmic_audio_map[] = {
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 1343ecb..6bb623a 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -39,7 +39,7 @@
 #define pcm_omap1510()	0
 #endif
 
-static const struct snd_pcm_hardware omap_pcm_hardware = {
+static struct snd_pcm_hardware omap_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_INTERLEAVED |
@@ -53,6 +53,24 @@
 	.buffer_bytes_max	= 128 * 1024,
 };
 
+/* sDMA supports only 1, 2, and 4 byte transfer elements. */
+static void omap_pcm_limit_supported_formats(void)
+{
+	int i;
+
+	for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
+		switch (snd_pcm_format_physical_width(i)) {
+		case 8:
+		case 16:
+		case 32:
+			omap_pcm_hardware.formats |= (1LL << i);
+			break;
+		default:
+			break;
+		}
+	}
+}
+
 /* this may get called several times by oss emulation */
 static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_pcm_hw_params *params)
@@ -235,6 +253,7 @@
 
 int omap_pcm_platform_register(struct device *dev)
 {
+	omap_pcm_limit_supported_formats();
 	return devm_snd_soc_register_platform(dev, &omap_soc_platform);
 }
 EXPORT_SYMBOL_GPL(omap_pcm_platform_register);
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index fb1f6bb..3673ada 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -170,14 +170,10 @@
 	if (priv->jack_detect > 0) {
 		hs_jack_gpios[0].gpio = priv->jack_detect;
 
-		ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
-				       &priv->hs_jack);
-		if (ret)
-			return ret;
-
-		ret = snd_soc_jack_add_pins(&priv->hs_jack,
-					    ARRAY_SIZE(hs_jack_pins),
-					    hs_jack_pins);
+		ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
+					    SND_JACK_HEADSET, &priv->hs_jack,
+					    hs_jack_pins,
+					    ARRAY_SIZE(hs_jack_pins));
 		if (ret)
 			return ret;
 
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 7f29935..c2ddf0f 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -311,9 +311,9 @@
 	}
 
 	/* AV jack detection */
-	err = snd_soc_jack_new(codec, "AV Jack",
-			       SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
-			       &rx51_av_jack);
+	err = snd_soc_card_jack_new(rtd->card, "AV Jack",
+				    SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
+				    &rx51_av_jack, NULL, 0);
 	if (err) {
 		dev_err(card->dev, "Failed to add AV Jack\n");
 		return err;
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 73eb5dd..9f8be7c 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -126,17 +126,12 @@
  */
 static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	/* Jack detection API stuff */
-	err = snd_soc_jack_new(codec, "Headphone Jack",
-				SND_JACK_HEADPHONE, &hs_jack);
-	if (err)
-		return err;
-
-	err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin),
-					hs_jack_pin);
+	err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin,
+				    ARRAY_SIZE(hs_jack_pin));
 	if (err)
 		return err;
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 910336c..c20bbc0 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -75,17 +75,12 @@
 
 static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	/* Jack detection API stuff */
-	err = snd_soc_jack_new(codec, "Headphone Jack",
-				SND_JACK_HEADPHONE, &hs_jack);
-	if (err)
-		return err;
-
-	err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-				hs_jack_pins);
+	err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins,
+				    ARRAY_SIZE(hs_jack_pins));
 	if (err)
 		return err;
 
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
index 5001dbb..1753c7d 100644
--- a/sound/soc/pxa/ttc-dkb.c
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -78,15 +78,12 @@
 	struct snd_soc_codec *codec = rtd->codec;
 
 	/* Headset jack detection */
-	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
-			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
-			&hs_jack);
-	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-			      hs_jack_pins);
-	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
-			 &mic_jack);
-	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
-			      mic_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE |
+			      SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			      &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
+	snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE,
+			      &mic_jack, mic_jack_pins,
+			      ARRAY_SIZE(mic_jack_pins));
 
 	/* headphone, microphone detection & headset short detection */
 	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 76ccb172..bcbfbe8 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -143,13 +143,9 @@
 	snd_soc_dapm_disable_pin(dapm, "MONO1");
 
 	/* Jack detection API stuff */
-	ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
-				&hs_jack);
-	if (ret)
-		goto err;
-
-	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
-				hs_jack_pins);
+	ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET,
+				    &hs_jack, hs_jack_pins,
+				    ARRAY_SIZE(hs_jack_pins));
 	if (ret)
 		goto err;
 
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 59b0442..c72e9fb 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -162,13 +162,8 @@
 
 static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
-
-	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
-		&hp_jack);
-
-	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
-		hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
+		&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
 
 	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
 		hp_jack_gpios);
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index 141519c..31a820e 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -260,12 +260,12 @@
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_jack_new(codec, "Headset",
-			       SND_JACK_HEADSET | SND_JACK_MECHANICAL |
-			       SND_JACK_BTN_0 | SND_JACK_BTN_1 |
-			       SND_JACK_BTN_2 | SND_JACK_BTN_3 |
-			       SND_JACK_BTN_4 | SND_JACK_BTN_5,
-			       &littlemill_headset);
+	ret = snd_soc_card_jack_new(card, "Headset",
+				    SND_JACK_HEADSET | SND_JACK_MECHANICAL |
+				    SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+				    SND_JACK_BTN_2 | SND_JACK_BTN_3 |
+				    SND_JACK_BTN_4 | SND_JACK_BTN_5,
+				    &littlemill_headset, NULL, 0);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c
index 243dea7..5f15609 100644
--- a/sound/soc/samsung/lowland.c
+++ b/sound/soc/samsung/lowland.c
@@ -56,16 +56,10 @@
 		return ret;
 	}
 
-	ret = snd_soc_jack_new(codec, "Headset",
-			       SND_JACK_LINEOUT | SND_JACK_HEADSET |
-			       SND_JACK_BTN_0,
-			       &lowland_headset);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(&lowland_headset,
-				    ARRAY_SIZE(lowland_headset_pins),
-				    lowland_headset_pins);
+	ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT |
+				    SND_JACK_HEADSET | SND_JACK_BTN_0,
+				    &lowland_headset, lowland_headset_pins,
+				    ARRAY_SIZE(lowland_headset_pins));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 873f2cb..35e37c4 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -211,13 +211,8 @@
 
 static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
-
-	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
-		&hp_jack);
-
-	snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
-		hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
+		&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
 
 	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
 		hp_jack_gpios);
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 8291d2a..dfbe2db 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -151,13 +151,10 @@
 	snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
 
 	/* Headphone jack detection */
-	err = snd_soc_jack_new(codec, "Headphone Jack",
-			       SND_JACK_HEADPHONE, &smartq_jack);
-	if (err)
-		return err;
-
-	err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins),
-				    smartq_jack_pins);
+	err = snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+				    SND_JACK_HEADPHONE, &smartq_jack,
+				    smartq_jack_pins,
+				    ARRAY_SIZE(smartq_jack_pins));
 	if (err)
 		return err;
 
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 5ec7c52..2dcb988 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -153,16 +153,10 @@
 		pr_err("Failed to request HP_SEL GPIO: %d\n", ret);
 	gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity);
 
-	ret = snd_soc_jack_new(codec, "Headset",
-			       SND_JACK_LINEOUT | SND_JACK_HEADSET |
-			       SND_JACK_BTN_0,
-			       &speyside_headset);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(&speyside_headset,
-				    ARRAY_SIZE(speyside_headset_pins),
-				    speyside_headset_pins);
+	ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT |
+				    SND_JACK_HEADSET | SND_JACK_BTN_0,
+				    &speyside_headset, speyside_headset_pins,
+				    ARRAY_SIZE(speyside_headset_pins));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c
index 9c80506..85ccfb7 100644
--- a/sound/soc/samsung/tobermory.c
+++ b/sound/soc/samsung/tobermory.c
@@ -179,15 +179,10 @@
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_jack_new(codec, "Headset",
-			       SND_JACK_HEADSET | SND_JACK_BTN_0,
-			       &tobermory_headset);
-	if (ret)
-		return ret;
-
-	ret = snd_soc_jack_add_pins(&tobermory_headset,
-				    ARRAY_SIZE(tobermory_headset_pins),
-				    tobermory_headset_pins);
+	ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET |
+				    SND_JACK_BTN_0, &tobermory_headset,
+				    tobermory_headset_pins,
+				    ARRAY_SIZE(tobermory_headset_pins));
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 30579ca..5c0658d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1561,6 +1561,10 @@
 		snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
 					  card->num_dapm_widgets);
 
+	if (card->of_dapm_widgets)
+		snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets,
+					  card->num_of_dapm_widgets);
+
 	/* initialise the sound card only once */
 	if (card->probe) {
 		ret = card->probe(card);
@@ -1616,6 +1620,10 @@
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 					card->num_dapm_routes);
 
+	if (card->of_dapm_routes)
+		snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
+					card->num_of_dapm_routes);
+
 	for (i = 0; i < card->num_links; i++) {
 		if (card->dai_link[i].dai_fmt)
 			snd_soc_runtime_set_dai_fmt(&card->rtd[i],
@@ -3223,8 +3231,8 @@
 		widgets[i].name = wname;
 	}
 
-	card->dapm_widgets = widgets;
-	card->num_dapm_widgets = num_widgets;
+	card->of_dapm_widgets = widgets;
+	card->num_of_dapm_widgets = num_widgets;
 
 	return 0;
 }
@@ -3308,8 +3316,8 @@
 		}
 	}
 
-	card->num_dapm_routes = num_routes;
-	card->dapm_routes = routes;
+	card->num_of_dapm_routes = num_routes;
+	card->of_dapm_routes = routes;
 
 	return 0;
 }
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 4380dcc..9f60c25 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -22,30 +22,42 @@
 #include <trace/events/asoc.h>
 
 /**
- * snd_soc_jack_new - Create a new jack
- * @codec: ASoC codec
+ * snd_soc_card_jack_new - Create a new jack
+ * @card:  ASoC card
  * @id:    an identifying string for this jack
  * @type:  a bitmask of enum snd_jack_type values that can be detected by
  *         this jack
  * @jack:  structure to use for the jack
+ * @pins:  Array of jack pins to be added to the jack or NULL
+ * @num_pins: Number of elements in the @pins array
  *
  * Creates a new jack object.
  *
  * Returns zero if successful, or a negative error code on failure.
  * On success jack will be initialised.
  */
-int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
-		     struct snd_soc_jack *jack)
+int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type,
+	struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins,
+	unsigned int num_pins)
 {
+	int ret;
+
 	mutex_init(&jack->mutex);
-	jack->codec = codec;
+	jack->card = card;
 	INIT_LIST_HEAD(&jack->pins);
 	INIT_LIST_HEAD(&jack->jack_zones);
 	BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier);
 
-	return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack);
+	ret = snd_jack_new(card->snd_card, id, type, &jack->jack);
+	if (ret)
+		return ret;
+
+	if (num_pins)
+		return snd_soc_jack_add_pins(jack, num_pins, pins);
+
+	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_soc_jack_new);
+EXPORT_SYMBOL_GPL(snd_soc_card_jack_new);
 
 /**
  * snd_soc_jack_report - Report the current status for a jack
@@ -63,7 +75,6 @@
  */
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 {
-	struct snd_soc_codec *codec;
 	struct snd_soc_dapm_context *dapm;
 	struct snd_soc_jack_pin *pin;
 	unsigned int sync = 0;
@@ -74,8 +85,7 @@
 	if (!jack)
 		return;
 
-	codec = jack->codec;
-	dapm =  &codec->dapm;
+	dapm = &jack->card->dapm;
 
 	mutex_lock(&jack->mutex);
 
@@ -175,12 +185,12 @@
 
 	for (i = 0; i < count; i++) {
 		if (!pins[i].pin) {
-			dev_err(jack->codec->dev, "ASoC: No name for pin %d\n",
+			dev_err(jack->card->dev, "ASoC: No name for pin %d\n",
 				i);
 			return -EINVAL;
 		}
 		if (!pins[i].mask) {
-			dev_err(jack->codec->dev, "ASoC: No mask for pin %d"
+			dev_err(jack->card->dev, "ASoC: No mask for pin %d"
 				" (%s)\n", i, pins[i].pin);
 			return -EINVAL;
 		}
@@ -260,7 +270,7 @@
 static irqreturn_t gpio_handler(int irq, void *data)
 {
 	struct snd_soc_jack_gpio *gpio = data;
-	struct device *dev = gpio->jack->codec->component.card->dev;
+	struct device *dev = gpio->jack->card->dev;
 
 	trace_snd_soc_jack_irq(gpio->name);
 
@@ -299,7 +309,7 @@
 
 	for (i = 0; i < count; i++) {
 		if (!gpios[i].name) {
-			dev_err(jack->codec->dev,
+			dev_err(jack->card->dev,
 				"ASoC: No name for gpio at index %d\n", i);
 			ret = -EINVAL;
 			goto undo;
@@ -320,7 +330,7 @@
 		} else {
 			/* legacy GPIO number */
 			if (!gpio_is_valid(gpios[i].gpio)) {
-				dev_err(jack->codec->dev,
+				dev_err(jack->card->dev,
 					"ASoC: Invalid gpio %d\n",
 					gpios[i].gpio);
 				ret = -EINVAL;
@@ -350,7 +360,7 @@
 		if (gpios[i].wake) {
 			ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1);
 			if (ret != 0)
-				dev_err(jack->codec->dev,
+				dev_err(jack->card->dev,
 					"ASoC: Failed to mark GPIO at index %d as wake source: %d\n",
 					i, ret);
 		}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 6b0136e..6e3781e 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2511,6 +2511,7 @@
 	/* DAPM dai link stream work */
 	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
+	pcm->nonatomic = rtd->dai_link->nonatomic;
 	rtd->pcm = pcm;
 	pcm->private_data = rtd;
 
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 769aca2..6dcd06a 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -106,11 +106,10 @@
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card);
 
-	snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
-			 &tegra_alc5632_hs_jack);
-	snd_soc_jack_add_pins(&tegra_alc5632_hs_jack,
-			ARRAY_SIZE(tegra_alc5632_hs_jack_pins),
-			tegra_alc5632_hs_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET,
+			      &tegra_alc5632_hs_jack,
+			      tegra_alc5632_hs_jack_pins,
+			      ARRAY_SIZE(tegra_alc5632_hs_jack_pins));
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det;
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index af3fb99..902da36 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -133,24 +133,26 @@
 	SND_SOC_DAPM_HP("Headphones", NULL),
 	SND_SOC_DAPM_SPK("Speakers", NULL),
 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
 };
 
 static const struct snd_kcontrol_new tegra_max98090_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphones"),
 	SOC_DAPM_PIN_SWITCH("Speakers"),
+	SOC_DAPM_PIN_SWITCH("Mic Jack"),
+	SOC_DAPM_PIN_SWITCH("Int Mic"),
 };
 
 static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = codec_dai->codec;
 	struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card);
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
-		snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE,
-				&tegra_max98090_hp_jack);
-		snd_soc_jack_add_pins(&tegra_max98090_hp_jack,
-				ARRAY_SIZE(tegra_max98090_hp_jack_pins),
-				tegra_max98090_hp_jack_pins);
+		snd_soc_card_jack_new(rtd->card, "Headphones",
+				      SND_JACK_HEADPHONE,
+				      &tegra_max98090_hp_jack,
+				      tegra_max98090_hp_jack_pins,
+				      ARRAY_SIZE(tegra_max98090_hp_jack_pins));
 
 		tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det;
 		snd_soc_jack_add_gpios(&tegra_max98090_hp_jack,
@@ -159,11 +161,11 @@
 	}
 
 	if (gpio_is_valid(machine->gpio_mic_det)) {
-		snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
-				 &tegra_max98090_mic_jack);
-		snd_soc_jack_add_pins(&tegra_max98090_mic_jack,
-				      ARRAY_SIZE(tegra_max98090_mic_jack_pins),
-				      tegra_max98090_mic_jack_pins);
+		snd_soc_card_jack_new(rtd->card, "Mic Jack",
+				      SND_JACK_MICROPHONE,
+				      &tegra_max98090_mic_jack,
+				      tegra_max98090_mic_jack_pins,
+				      ARRAY_SIZE(tegra_max98090_mic_jack_pins));
 
 		tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det;
 		snd_soc_jack_add_gpios(&tegra_max98090_mic_jack,
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index ed759a3..773daec 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -108,15 +108,11 @@
 
 static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = codec_dai->codec;
 	struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card);
 
-	snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE,
-			 &tegra_rt5640_hp_jack);
-	snd_soc_jack_add_pins(&tegra_rt5640_hp_jack,
-			ARRAY_SIZE(tegra_rt5640_hp_jack_pins),
-			tegra_rt5640_hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE,
+			      &tegra_rt5640_hp_jack, tegra_rt5640_hp_jack_pins,
+			      ARRAY_SIZE(tegra_rt5640_hp_jack_pins));
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det;
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index e4cf978..68d8b67 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -146,10 +146,9 @@
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card);
 
-	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
-			&tegra_rt5677_hp_jack);
-	snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1,
-			&tegra_rt5677_hp_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
+			      &tegra_rt5677_hp_jack,
+			      &tegra_rt5677_hp_jack_pins, 1);
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det;
@@ -158,10 +157,9 @@
 	}
 
 
-	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
-			&tegra_rt5677_mic_jack);
-	snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1,
-			&tegra_rt5677_mic_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
+			      &tegra_rt5677_mic_jack,
+			      &tegra_rt5677_mic_jack_pins, 1);
 
 	if (gpio_is_valid(machine->gpio_mic_present)) {
 		tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present;
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index e52420d..4a95b70 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -177,21 +177,19 @@
 
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det;
-		snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
-				&tegra_wm8903_hp_jack);
-		snd_soc_jack_add_pins(&tegra_wm8903_hp_jack,
-					ARRAY_SIZE(tegra_wm8903_hp_jack_pins),
-					tegra_wm8903_hp_jack_pins);
+		snd_soc_card_jack_new(rtd->card, "Headphone Jack",
+				      SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack,
+				      tegra_wm8903_hp_jack_pins,
+				      ARRAY_SIZE(tegra_wm8903_hp_jack_pins));
 		snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
 					1,
 					&tegra_wm8903_hp_jack_gpio);
 	}
 
-	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
-			 &tegra_wm8903_mic_jack);
-	snd_soc_jack_add_pins(&tegra_wm8903_mic_jack,
-			      ARRAY_SIZE(tegra_wm8903_mic_jack_pins),
-			      tegra_wm8903_mic_jack_pins);
+	snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
+			      &tegra_wm8903_mic_jack,
+			      tegra_wm8903_mic_jack_pins,
+			      ARRAY_SIZE(tegra_wm8903_mic_jack_pins));
 	wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE,
 				0);
 
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 753a47d..353532b 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1120,17 +1120,24 @@
 /* Marantz/Denon USB DACs need a vendor cmd to switch
  * between PCM and native DSD mode
  */
+static bool is_marantz_denon_dac(unsigned int id)
+{
+	switch (id) {
+	case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
+	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
+	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
+		return true;
+	}
+	return false;
+}
+
 int snd_usb_select_mode_quirk(struct snd_usb_substream *subs,
 			      struct audioformat *fmt)
 {
 	struct usb_device *dev = subs->dev;
 	int err;
 
-	switch (subs->stream->chip->usb_id) {
-	case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
-	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
-	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
-
+	if (is_marantz_denon_dac(subs->stream->chip->usb_id)) {
 		/* First switch to alt set 0, otherwise the mode switch cmd
 		 * will not be accepted by the DAC
 		 */
@@ -1203,17 +1210,10 @@
 	/* Marantz/Denon devices with USB DAC functionality need a delay
 	 * after each class compliant request
 	 */
-	if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) &&
-	    (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) {
-
-		switch (le16_to_cpu(dev->descriptor.idProduct)) {
-		case 0x1003: /* Denon DA300-USB */
-		case 0x3005: /* Marantz HD-DAC1 */
-		case 0x3006: /* Marantz SA-14S1 */
-			mdelay(20);
-			break;
-		}
-	}
+	if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor),
+					le16_to_cpu(dev->descriptor.idProduct)))
+	    && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+		mdelay(20);
 
 	/* Zoom R16/24 needs a tiny delay here, otherwise requests like
 	 * get/set frequency return as failed despite actually succeeding.
@@ -1268,15 +1268,9 @@
 	}
 
 	/* Denon/Marantz devices with USB DAC functionality */
-	switch (chip->usb_id) {
-	case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */
-	case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
-	case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
+	if (is_marantz_denon_dac(chip->usb_id)) {
 		if (fp->altsetting == 2)
 			return SNDRV_PCM_FMTBIT_DSD_U32_BE;
-		break;
-	default:
-		break;
 	}
 
 	return 0;