Merge branch 'topic/timestamp' into for-next

7 files changed, 444 insertions, 55 deletions

diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
new file mode 100644
index 000000000000..0b191a23f534
--- /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/pcm.h b/include/sound/pcm.h
index c0ddb7e69c28..0cb7f3f5df7b 100644
--- a/include/sound/pcm.h
+++ b/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;
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index fd5c0b55ca3b..46145a5277fe 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -143,7 +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;
@@ -270,10 +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 */
@@ -411,6 +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 */
@@ -422,9 +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 {
@@ -537,6 +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)
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index 2d957ba63557..b48b434444ed 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -194,18 +194,30 @@ struct snd_pcm_status32 {
 	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 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream,
 	    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 @@ enum {
 	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 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l
 	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_lib.c b/sound/core/pcm_lib.c
index ffd656012ab8..ac6b33f3779c 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -232,6 +232,49 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream,
 	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 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	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 ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) &&
-			(substream->ops->wall_clock))
-			substream->ops->wall_clock(substream, &audio_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);
+
+			/* 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 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	}
 
  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 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		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 b03a638b420c..9c2c6f801fed 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -707,6 +707,23 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
 	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 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
 		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 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
 }
 
 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;
@@ -2723,7 +2756,9 @@ static int snd_pcm_common_ioctl1(struct file *file,
 	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/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 2a674525e56f..ebb7a644bd86 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -731,17 +731,32 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
 			       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;
 }
@@ -755,7 +770,8 @@ static struct snd_pcm_hardware azx_pcm_hw = {
 				 /* 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,
@@ -841,10 +857,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 		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;
@@ -876,7 +894,7 @@ static struct snd_pcm_ops azx_pcm_ops = {
 	.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,
 };