Commit a3a7d2e0 authored by Vivia Nikolaidou's avatar Vivia Nikolaidou 🦆

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
parent 42a03a81
......@@ -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 +705,48 @@ 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);
/* With drop-frame we have gaps of 1 sample every now and then (rounding
* errors because of the samples-per-frame pattern which is not 100%
* accurate), and due to rounding errors in the calculations these can be
* 2>x>1 */
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 +953,9 @@ 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,
......@@ -825,6 +829,29 @@ gst_decklink_video_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
}
}
/* 1 ns error can be just a rounding error, so that's OK. The Decklink
* drivers give us a really steady stream time, so anything above 1 ns can't
* be a rounding error and is therefore something to worry about */
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);
......@@ -1080,6 +1107,10 @@ 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;
......
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