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ALSA: core: selection of audio_tstamp type and accuracy reports 

Audio timestamps can be extracted from sample counters, wall clocks,
PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This
patch provides the ability to report timestamping capabilities, select
timestamp types and retrieve timestamp accuracy, if supported.
Details can be found in Documentations/sound/alsa/timestamping.txt

This functionality is introduced by reclaiming the reserved_aligned
field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a
in snd_pcm_status to provide userspace with selection/query capabilities.
Additional driver_tstamp and audio_tstamp_accuracy fields are also added.

snd_pcm_mmap_status remains a read-only structure with only
the audio timestamp value accessible from user space. The selection
of audio timestamp type is done through snd_pcm_status only

This commit does not impact ABI and does not impact the default
behavior. By default audio timestamp is aligned with hw_pointer and
reports the DMA position. Backwards compatibility is handled by using
the HDAudio wall clock for playback and the hw_ptr for all other
cases.

For timestamp selection a new STATUS_EXT ioctl is introduced with
read/write parameters. Alsa-lib will be modified to make use of
STATUS_EXT.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is contained in:
Pierre-Louis Bossart 2015-02-13 15:14:03 -06:00 committed by Takashi Iwai
parent 0c8d948565
commit 229d043096
3 changed files with 290 additions and 4 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

200
Documentation/sound/alsa/timestamping.txt Normal file
View File

@ -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

60
include/sound/pcm.h
View File

@ -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);
@ -281,6 +284,58 @@ 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 +416,11 @@ 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;

34
include/uapi/sound/asound.h
View File

@ -267,10 +267,17 @@ 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 +415,22 @@ 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 +442,11 @@ 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 +559,7 @@ 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)