mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
decklink: Detect gaps on incoming stream times, issue warnings
When we receive a video or audio buffer, we calculate the next stream time based on the current stream time + buffer duration. If the next buffer's stream time is after that, we issue a warning. This happens because the stream time incoming from Decklink should be really constant and without gaps. If there is a gap, it means that something went wrong, e.g. the internal buffer pool is empty (too many buffers queued up downstream). https://bugzilla.gnome.org/show_bug.cgi?id=781776
This commit is contained in:
parent
1cef7a261f
commit
c1294e10f9
4 changed files with 82 additions and 0 deletions
|
@ -36,6 +36,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_src_debug);
|
|||
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
|
||||
#define DEFAULT_CHANNELS (GST_DECKLINK_AUDIO_CHANNELS_2)
|
||||
|
||||
#ifndef ABSDIFF
|
||||
#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
|
||||
#endif
|
||||
|
||||
enum
|
||||
{
|
||||
PROP_0,
|
||||
|
@ -701,6 +705,44 @@ retry:
|
|||
self->info.rate) - timestamp;
|
||||
}
|
||||
|
||||
// Detect gaps in stream time
|
||||
self->processed += sample_count;
|
||||
|
||||
if (p.stream_timestamp != GST_CLOCK_TIME_NONE) {
|
||||
GstClockTime start_stream_time, end_stream_time;
|
||||
|
||||
start_stream_time = p.stream_timestamp;
|
||||
|
||||
start_offset =
|
||||
gst_util_uint64_scale (start_stream_time, self->info.rate, GST_SECOND);
|
||||
|
||||
end_offset = start_offset + sample_count;
|
||||
end_stream_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
|
||||
self->info.rate);
|
||||
|
||||
if (self->expected_stream_time != GST_CLOCK_TIME_NONE &&
|
||||
ABSDIFF (self->expected_stream_time, p.stream_timestamp) >
|
||||
gst_util_uint64_scale (2, GST_SECOND, self->info.rate)) {
|
||||
GstMessage *msg;
|
||||
GstClockTime running_time;
|
||||
|
||||
self->dropped +=
|
||||
gst_util_uint64_scale (ABSDIFF (self->expected_stream_time,
|
||||
p.stream_timestamp), self->info.rate, GST_SECOND);
|
||||
running_time =
|
||||
gst_segment_to_running_time (&GST_BASE_SRC (self)->segment,
|
||||
GST_FORMAT_TIME, timestamp);
|
||||
|
||||
msg =
|
||||
gst_message_new_qos (GST_OBJECT (self), TRUE, running_time, p.stream_timestamp,
|
||||
timestamp, duration);
|
||||
gst_message_set_qos_stats (msg, GST_FORMAT_DEFAULT, self->processed,
|
||||
self->dropped);
|
||||
gst_element_post_message (GST_ELEMENT (self), msg);
|
||||
}
|
||||
self->expected_stream_time = end_stream_time;
|
||||
}
|
||||
|
||||
if (p.no_signal)
|
||||
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_GAP);
|
||||
GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
|
||||
|
@ -907,6 +949,9 @@ gst_decklink_audio_src_change_state (GstElement * element,
|
|||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
self->processed = 0;
|
||||
self->dropped = 0;
|
||||
self->expected_stream_time = GST_CLOCK_TIME_NONE;
|
||||
if (!gst_decklink_audio_src_open (self)) {
|
||||
ret = GST_STATE_CHANGE_FAILURE;
|
||||
goto out;
|
||||
|
|
|
@ -71,6 +71,11 @@ struct _GstDecklinkAudioSrc
|
|||
/* counter to keep track of timestamps */
|
||||
guint64 next_offset;
|
||||
|
||||
/* detect gaps in stream time */
|
||||
GstClockTime expected_stream_time;
|
||||
guint64 processed;
|
||||
guint64 dropped;
|
||||
|
||||
/* Last time we noticed a discont */
|
||||
GstClockTime discont_time;
|
||||
|
||||
|
|
|
@ -36,6 +36,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_decklink_video_src_debug);
|
|||
#define DEFAULT_SKIP_FIRST_TIME (0)
|
||||
#define DEFAULT_DROP_NO_SIGNAL_FRAMES (FALSE)
|
||||
|
||||
#ifndef ABSDIFF
|
||||
#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
|
||||
#endif
|
||||
|
||||
enum
|
||||
{
|
||||
PROP_0,
|
||||
|
@ -832,6 +836,26 @@ gst_decklink_video_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
|
|||
}
|
||||
}
|
||||
|
||||
if (self->expected_stream_time != GST_CLOCK_TIME_NONE &&
|
||||
ABSDIFF (self->expected_stream_time, f.stream_timestamp) > 1) {
|
||||
GstMessage *msg;
|
||||
GstClockTime running_time;
|
||||
|
||||
self->dropped += f.stream_timestamp - self->expected_stream_time;
|
||||
running_time = gst_segment_to_running_time (&GST_BASE_SRC (self)->segment,
|
||||
GST_FORMAT_TIME, f.timestamp);
|
||||
|
||||
msg = gst_message_new_qos (GST_OBJECT (self), TRUE, running_time, f.stream_timestamp,
|
||||
f.timestamp, f.duration);
|
||||
gst_message_set_qos_stats (msg, GST_FORMAT_TIME, self->processed,
|
||||
self->dropped);
|
||||
gst_element_post_message (GST_ELEMENT (self), msg);
|
||||
}
|
||||
if (self->first_stream_time == GST_CLOCK_TIME_NONE)
|
||||
self->first_stream_time = f.stream_timestamp;
|
||||
self->processed = f.stream_timestamp - self->dropped - self->first_stream_time;
|
||||
self->expected_stream_time = f.stream_timestamp + f.stream_duration;
|
||||
|
||||
g_mutex_unlock (&self->lock);
|
||||
if (caps_changed) {
|
||||
caps = gst_decklink_mode_get_caps (f.mode, f.format, TRUE);
|
||||
|
@ -1087,6 +1111,10 @@ gst_decklink_video_src_change_state (GstElement * element,
|
|||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
self->processed = 0;
|
||||
self->dropped = 0;
|
||||
self->expected_stream_time = GST_CLOCK_TIME_NONE;
|
||||
self->first_stream_time = GST_CLOCK_TIME_NONE;
|
||||
if (!gst_decklink_video_src_open (self)) {
|
||||
ret = GST_STATE_CHANGE_FAILURE;
|
||||
goto out;
|
||||
|
|
|
@ -58,6 +58,10 @@ struct _GstDecklinkVideoSrc
|
|||
gboolean output_stream_time;
|
||||
GstClockTime skip_first_time;
|
||||
gboolean drop_no_signal_frames;
|
||||
GstClockTime expected_stream_time;
|
||||
guint64 processed;
|
||||
guint64 dropped;
|
||||
guint64 first_stream_time;
|
||||
|
||||
GstVideoInfo info;
|
||||
GstDecklinkVideoFormat video_format;
|
||||
|
|
Loading…
Reference in a new issue