Merge branch 'for-next' into for-linus

Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt

@@ -18,6 +18,7 @@ Required properties:
   * Headphones
   * Speakers
   * Mic Jack
+  * Int Mic
 
 - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
   connected to the CODEC.

Documentation/sound/alsa/HD-Audio.txt

@@ -466,7 +466,11 @@ The generic parser supports the following hints:
 - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each
   I/O jack for allowing to change the headphone amp and mic bias VREF
   capabilities
-- power_down_unused (bool): power down the unused widgets
+- power_save_node (bool): advanced power management for each widget,
+  controlling the power sate (D0/D3) of each widget node depending on
+  the actual pin and stream states
+- power_down_unused (bool): power down the unused widgets, a subset of
+  power_save_node, and will be dropped in future
 - add_hp_mic (bool): add the headphone to capture source if possible
 - hp_mic_detect (bool): enable/disable the hp/mic shared input for a
   single built-in mic case; default true

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

include/sound/ac97_codec.h

@@ -608,7 +608,9 @@ struct ac97_quirk {
 	int type;		/* quirk type above */
 };
 
-int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override);
+int snd_ac97_tune_hardware(struct snd_ac97 *ac97,
+			   const struct ac97_quirk *quirk,
+			   const char *override);
 int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate);
 
 /*

include/sound/compress_driver.h

@@ -70,7 +70,7 @@ struct snd_compr_runtime {
  * @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 @@ struct snd_compr_stream {
  * 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

include/sound/control.h

@@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
  * 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,

include/sound/core.h

@@ -278,7 +278,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type,
 		   void *device_data, struct snd_device_ops *ops);
 int snd_device_register(struct snd_card *card, void *device_data);
 int snd_device_register_all(struct snd_card *card);
-int snd_device_disconnect_all(struct snd_card *card);
+void snd_device_disconnect(struct snd_card *card, void *device_data);
+void snd_device_disconnect_all(struct snd_card *card);
 void snd_device_free(struct snd_card *card, void *device_data);
 void snd_device_free_all(struct snd_card *card);
 

include/sound/hda_regmap.h

@@ -0,0 +1,217 @@
+/*
+ * HD-audio regmap helpers
+ */
+
+#ifndef __SOUND_HDA_REGMAP_H
+#define __SOUND_HDA_REGMAP_H
+
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/hdaudio.h>
+
+int snd_hdac_regmap_init(struct hdac_device *codec);
+void snd_hdac_regmap_exit(struct hdac_device *codec);
+int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec,
+				    unsigned int verb);
+int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg,
+			     unsigned int *val);
+int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg,
+			      unsigned int val);
+int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg,
+			       unsigned int mask, unsigned int val);
+
+/**
+ * snd_hdac_regmap_encode_verb - encode the verb to a pseudo register
+ * @nid: widget NID
+ * @verb: codec verb
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_verb(nid, verb)		\
+	(((verb) << 8) | 0x80000 | ((unsigned int)(nid) << 20))
+
+/**
+ * snd_hdac_regmap_encode_amp - encode the AMP verb to a pseudo register
+ * @nid: widget NID
+ * @ch: channel (left = 0, right = 1)
+ * @dir: direction (#HDA_INPUT, #HDA_OUTPUT)
+ * @idx: input index value
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_amp(nid, ch, dir, idx)			\
+	(snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) |	\
+	 ((ch) ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT) |			\
+	 ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \
+	 (idx))
+
+/**
+ * snd_hdac_regmap_encode_amp_stereo - encode a pseudo register for stereo AMPs
+ * @nid: widget NID
+ * @dir: direction (#HDA_INPUT, #HDA_OUTPUT)
+ * @idx: input index value
+ *
+ * Returns an encoded pseudo register.
+ */
+#define snd_hdac_regmap_encode_amp_stereo(nid, dir, idx)		\
+	(snd_hdac_regmap_encode_verb(nid, AC_VERB_GET_AMP_GAIN_MUTE) |	\
+	 AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT | /* both bits set! */	\
+	 ((dir) == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT) | \
+	 (idx))
+
+/**
+ * snd_hdac_regmap_write - Write a verb with caching
+ * @nid: codec NID
+ * @reg: verb to write
+ * @val: value to write
+ *
+ * For writing an amp value, use snd_hda_regmap_amp_update().
+ */
+static inline int
+snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid,
+		      unsigned int verb, unsigned int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+	return snd_hdac_regmap_write_raw(codec, cmd, val);
+}
+
+/**
+ * snd_hda_regmap_update - Update a verb value with caching
+ * @nid: codec NID
+ * @verb: verb to update
+ * @mask: bit mask to update
+ * @val: value to update
+ *
+ * For updating an amp value, use snd_hda_regmap_amp_update().
+ */
+static inline int
+snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid,
+		       unsigned int verb, unsigned int mask,
+		       unsigned int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+	return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hda_regmap_read - Read a verb with caching
+ * @nid: codec NID
+ * @verb: verb to read
+ * @val: pointer to store the value
+ *
+ * For reading an amp value, use snd_hda_regmap_get_amp().
+ */
+static inline int
+snd_hdac_regmap_read(struct hdac_device *codec, hda_nid_t nid,
+		     unsigned int verb, unsigned int *val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_verb(nid, verb);
+
+	return snd_hdac_regmap_read_raw(codec, cmd, val);
+}
+
+/**
+ * snd_hdac_regmap_get_amp - Read AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ * @val: the pointer to store the value
+ *
+ * Read AMP value.  The volume is between 0 to 0x7f, 0x80 = mute bit.
+ * Returns the value or a negative error.
+ */
+static inline int
+snd_hdac_regmap_get_amp(struct hdac_device *codec, hda_nid_t nid,
+			int ch, int dir, int idx)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+	int err, val;
+
+	err = snd_hdac_regmap_read_raw(codec, cmd, &val);
+	return err < 0 ? err : val;
+}
+
+/**
+ * snd_hdac_regmap_update_amp - update the AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the AMP value with a bit mask.
+ * Returns 0 if the value is unchanged, 1 if changed, or a negative error.
+ */
+static inline int
+snd_hdac_regmap_update_amp(struct hdac_device *codec, hda_nid_t nid,
+			   int ch, int dir, int idx, int mask, int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+
+	return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hdac_regmap_get_amp_stereo - Read stereo AMP values
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ * @val: the pointer to store the value
+ *
+ * Read stereo AMP values.  The lower byte is left, the upper byte is right.
+ * Returns the value or a negative error.
+ */
+static inline int
+snd_hdac_regmap_get_amp_stereo(struct hdac_device *codec, hda_nid_t nid,
+			       int dir, int idx)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx);
+	int err, val;
+
+	err = snd_hdac_regmap_read_raw(codec, cmd, &val);
+	return err < 0 ? err : val;
+}
+
+/**
+ * snd_hdac_regmap_update_amp_stereo - update the stereo AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the stereo AMP value with a bit mask.
+ * The lower byte is left, the upper byte is right.
+ * Returns 0 if the value is unchanged, 1 if changed, or a negative error.
+ */
+static inline int
+snd_hdac_regmap_update_amp_stereo(struct hdac_device *codec, hda_nid_t nid,
+				  int dir, int idx, int mask, int val)
+{
+	unsigned int cmd = snd_hdac_regmap_encode_amp_stereo(nid, dir, idx);
+
+	return snd_hdac_regmap_update_raw(codec, cmd, mask, val);
+}
+
+/**
+ * snd_hdac_regmap_sync_node - sync the widget node attributes
+ * @codec: HD-audio codec
+ * @nid: NID to sync
+ */
+static inline void
+snd_hdac_regmap_sync_node(struct hdac_device *codec, hda_nid_t nid)
+{
+	regcache_mark_dirty(codec->regmap);
+	regcache_sync_region(codec->regmap, nid << 20, ((nid + 1) << 20) - 1);
+}
+
+#endif /* __SOUND_HDA_REGMAP_H */

include/sound/hdaudio.h

@@ -0,0 +1,271 @@
+/*
+ * HD-audio core stuff
+ */
+
+#ifndef __SOUND_HDAUDIO_H
+#define __SOUND_HDAUDIO_H
+
+#include <linux/device.h>
+#include <sound/hda_verbs.h>
+
+/* codec node id */
+typedef u16 hda_nid_t;
+
+struct hdac_bus;
+struct hdac_device;
+struct hdac_driver;
+struct hdac_widget_tree;
+
+/*
+ * exported bus type
+ */
+extern struct bus_type snd_hda_bus_type;
+
+/*
+ * generic arrays
+ */
+struct snd_array {
+	unsigned int used;
+	unsigned int alloced;
+	unsigned int elem_size;
+	unsigned int alloc_align;
+	void *list;
+};
+
+/*
+ * HD-audio codec base device
+ */
+struct hdac_device {
+	struct device dev;
+	int type;
+	struct hdac_bus *bus;
+	unsigned int addr;		/* codec address */
+	struct list_head list;		/* list point for bus codec_list */
+
+	hda_nid_t afg;			/* AFG node id */
+	hda_nid_t mfg;			/* MFG node id */
+
+	/* ids */
+	unsigned int vendor_id;
+	unsigned int subsystem_id;
+	unsigned int revision_id;
+	unsigned int afg_function_id;
+	unsigned int mfg_function_id;
+	unsigned int afg_unsol:1;
+	unsigned int mfg_unsol:1;
+
+	unsigned int power_caps;	/* FG power caps */
+
+	const char *vendor_name;	/* codec vendor name */
+	const char *chip_name;		/* codec chip name */
+
+	/* verb exec op override */
+	int (*exec_verb)(struct hdac_device *dev, unsigned int cmd,
+			 unsigned int flags, unsigned int *res);
+
+	/* widgets */
+	unsigned int num_nodes;
+	hda_nid_t start_nid, end_nid;
+
+	/* misc flags */
+	atomic_t in_pm;		/* suspend/resume being performed */
+
+	/* sysfs */
+	struct hdac_widget_tree *widgets;
+
+	/* regmap */
+	struct regmap *regmap;
+	struct snd_array vendor_verbs;
+	bool lazy_cache:1;	/* don't wake up for writes */
+	bool caps_overwriting:1; /* caps overwrite being in process */
+	bool cache_coef:1;	/* cache COEF read/write too */
+};
+
+/* device/driver type used for matching */
+enum {
+	HDA_DEV_CORE,
+	HDA_DEV_LEGACY,
+};
+
+/* direction */
+enum {
+	HDA_INPUT, HDA_OUTPUT
+};
+
+#define dev_to_hdac_dev(_dev)	container_of(_dev, struct hdac_device, dev)
+
+int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus,
+			 const char *name, unsigned int addr);
+void snd_hdac_device_exit(struct hdac_device *dev);
+int snd_hdac_device_register(struct hdac_device *codec);
+void snd_hdac_device_unregister(struct hdac_device *codec);
+
+int snd_hdac_refresh_widgets(struct hdac_device *codec);
+
+unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
+			       unsigned int verb, unsigned int parm);
+int snd_hdac_exec_verb(struct hdac_device *codec, unsigned int cmd,
+		       unsigned int flags, unsigned int *res);
+int snd_hdac_read(struct hdac_device *codec, hda_nid_t nid,
+		  unsigned int verb, unsigned int parm, unsigned int *res);
+int _snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm,
+			unsigned int *res);
+int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid,
+				int parm);
+int snd_hdac_override_parm(struct hdac_device *codec, hda_nid_t nid,
+			   unsigned int parm, unsigned int val);
+int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid,
+			     hda_nid_t *conn_list, int max_conns);
+int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid,
+			   hda_nid_t *start_id);
+
+/**
+ * snd_hdac_read_parm - read a codec parameter
+ * @codec: the codec object
+ * @nid: NID to read a parameter
+ * @parm: parameter to read
+ *
+ * Returns -1 for error.  If you need to distinguish the error more
+ * strictly, use _snd_hdac_read_parm() directly.
+ */
+static inline int snd_hdac_read_parm(struct hdac_device *codec, hda_nid_t nid,
+				     int parm)
+{
+	unsigned int val;
+
+	return _snd_hdac_read_parm(codec, nid, parm, &val) < 0 ? -1 : val;
+}
+
+#ifdef CONFIG_PM
+void snd_hdac_power_up(struct hdac_device *codec);
+void snd_hdac_power_down(struct hdac_device *codec);
+#else
+static inline void snd_hdac_power_up(struct hdac_device *codec) {}
+static inline void snd_hdac_power_down(struct hdac_device *codec) {}
+#endif
+
+/**
+ * snd_hdac_power_up_pm - power up the codec
+ * @codec: the codec object
+ *
+ * This function can be called in a recursive code path like init code
+ * which may be called by PM suspend/resume again.  OTOH, if a power-up
+ * call must wake up the sleeper (e.g. in a kctl callback), use
+ * snd_hdac_power_up() instead.
+ */
+static inline void snd_hdac_power_up_pm(struct hdac_device *codec)
+{
+	if (!atomic_read(&codec->in_pm))
+		snd_hdac_power_up(codec);
+}
+
+/**
+ * snd_hdac_power_down_pm - power down the codec
+ * @codec: the codec object
+ *
+ * Like snd_hdac_power_up_pm(), this function is used in a recursive
+ * code path like init code which may be called by PM suspend/resume again.
+ */
+static inline void snd_hdac_power_down_pm(struct hdac_device *codec)
+{
+	if (!atomic_read(&codec->in_pm))
+		snd_hdac_power_down(codec);
+}
+
+/*
+ * HD-audio codec base driver
+ */
+struct hdac_driver {
+	struct device_driver driver;
+	int type;
+	int (*match)(struct hdac_device *dev, struct hdac_driver *drv);
+	void (*unsol_event)(struct hdac_device *dev, unsigned int event);
+};
+
+#define drv_to_hdac_driver(_drv) container_of(_drv, struct hdac_driver, driver)
+
+/*
+ * HD-audio bus base driver
+ */
+struct hdac_bus_ops {
+	/* send a single command */
+	int (*command)(struct hdac_bus *bus, unsigned int cmd);
+	/* get a response from the last command */
+	int (*get_response)(struct hdac_bus *bus, unsigned int addr,
+			    unsigned int *res);
+};
+
+#define HDA_UNSOL_QUEUE_SIZE	64
+
+struct hdac_bus {
+	struct device *dev;
+	const struct hdac_bus_ops *ops;
+
+	/* codec linked list */
+	struct list_head codec_list;
+	unsigned int num_codecs;
+
+	/* link caddr -> codec */
+	struct hdac_device *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
+
+	/* unsolicited event queue */
+	u32 unsol_queue[HDA_UNSOL_QUEUE_SIZE * 2]; /* ring buffer */
+	unsigned int unsol_rp, unsol_wp;
+	struct work_struct unsol_work;
+
+	/* bit flags of powered codecs */
+	unsigned long codec_powered;
+
+	/* flags */
+	bool sync_write:1;		/* sync after verb write */
+
+	/* locks */
+	struct mutex cmd_mutex;
+};
+
+int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
+		      const struct hdac_bus_ops *ops);
+void snd_hdac_bus_exit(struct hdac_bus *bus);
+int snd_hdac_bus_exec_verb(struct hdac_bus *bus, unsigned int addr,
+			   unsigned int cmd, unsigned int *res);
+int snd_hdac_bus_exec_verb_unlocked(struct hdac_bus *bus, unsigned int addr,
+				    unsigned int cmd, unsigned int *res);
+void snd_hdac_bus_queue_event(struct hdac_bus *bus, u32 res, u32 res_ex);
+
+int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec);
+void snd_hdac_bus_remove_device(struct hdac_bus *bus,
+				struct hdac_device *codec);
+
+static inline void snd_hdac_codec_link_up(struct hdac_device *codec)
+{
+	set_bit(codec->addr, &codec->bus->codec_powered);
+}
+
+static inline void snd_hdac_codec_link_down(struct hdac_device *codec)
+{
+	clear_bit(codec->addr, &codec->bus->codec_powered);
+}
+
+/*
+ * generic array helpers
+ */
+void *snd_array_new(struct snd_array *array);
+void snd_array_free(struct snd_array *array);
+static inline void snd_array_init(struct snd_array *array, unsigned int size,
+				  unsigned int align)
+{
+	array->elem_size = size;
+	array->alloc_align = align;
+}
+
+static inline void *snd_array_elem(struct snd_array *array, unsigned int idx)
+{
+	return array->list + idx * array->elem_size;
+}
+
+static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
+{
+	return (unsigned long)(ptr - array->list) / array->elem_size;
+}
+
+#endif /* __SOUND_HDAUDIO_H */

include/sound/pcm.h

@@ -60,6 +60,9 @@ struct snd_pcm_hardware {
 
 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 @@ struct snd_pcm_ops {
 	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_hw_constraint_ranges {
 
 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_pcm_runtime {
 
 	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;

include/sound/pcm_params.h

@@ -366,4 +366,11 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p)
 	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 */

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 @@ struct snd_seq_device {
  *		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

include/sound/seq_kernel.h

@@ -99,13 +99,9 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp,
 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

include/sound/soc.h

@@ -450,8 +450,10 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream,
 		      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_gpio {
 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 @@ struct snd_soc_dai_link {
 	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 @@ struct snd_soc_card {
 
 	/*
 	 * 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 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec(
 }
 
 /**
- * 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

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)

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 @@ struct snd_hwdep_dsp_image {
  *                                                                           *
  *****************************************************************************/
 
-#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 @@ typedef int __bitwise snd_pcm_subformat_t;
 #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 @@ struct snd_pcm_channel_info {
 	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 @@ struct snd_pcm_status {
 	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 @@ enum {
 #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)

include/uapi/sound/compress_offload.h

@@ -75,7 +75,7 @@ struct snd_compr_tstamp {
 /**
  * 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;

include/uapi/sound/emu10k1.h

@@ -23,8 +23,7 @@
 #define _UAPI__SOUND_EMU10K1_H
 
 #include <linux/types.h>
-
-
+#include <sound/asound.h>
 
 /*
  * ---- FX8010 ----

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
 

sound/Kconfig

@@ -76,6 +76,8 @@ source "sound/isa/Kconfig"
 
 source "sound/pci/Kconfig"
 
+source "sound/hda/Kconfig"
+
 source "sound/ppc/Kconfig"
 
 source "sound/aoa/Kconfig"