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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
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4 changed files with 89 additions and 0 deletions
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@ -36,6 +36,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_src_debug);
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#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
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#define DEFAULT_CHANNELS (GST_DECKLINK_AUDIO_CHANNELS_2)
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#ifndef ABSDIFF
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#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
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#endif
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enum
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{
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PROP_0,
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@ -701,6 +705,48 @@ retry:
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self->info.rate) - timestamp;
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}
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// Detect gaps in stream time
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self->processed += sample_count;
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if (p.stream_timestamp != GST_CLOCK_TIME_NONE) {
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GstClockTime start_stream_time, end_stream_time;
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start_stream_time = p.stream_timestamp;
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start_offset =
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gst_util_uint64_scale (start_stream_time, self->info.rate, GST_SECOND);
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end_offset = start_offset + sample_count;
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end_stream_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
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self->info.rate);
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/* With drop-frame we have gaps of 1 sample every now and then (rounding
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* errors because of the samples-per-frame pattern which is not 100%
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* accurate), and due to rounding errors in the calculations these can be
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* 2>x>1 */
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if (self->expected_stream_time != GST_CLOCK_TIME_NONE &&
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ABSDIFF (self->expected_stream_time, p.stream_timestamp) >
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gst_util_uint64_scale (2, GST_SECOND, self->info.rate)) {
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GstMessage *msg;
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GstClockTime running_time;
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self->dropped +=
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gst_util_uint64_scale (ABSDIFF (self->expected_stream_time,
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p.stream_timestamp), self->info.rate, GST_SECOND);
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running_time =
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gst_segment_to_running_time (&GST_BASE_SRC (self)->segment,
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GST_FORMAT_TIME, timestamp);
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msg =
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gst_message_new_qos (GST_OBJECT (self), TRUE, running_time, p.stream_timestamp,
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timestamp, duration);
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gst_message_set_qos_stats (msg, GST_FORMAT_DEFAULT, self->processed,
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self->dropped);
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gst_element_post_message (GST_ELEMENT (self), msg);
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}
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self->expected_stream_time = end_stream_time;
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}
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if (p.no_signal)
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GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_GAP);
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GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
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@ -907,6 +953,9 @@ gst_decklink_audio_src_change_state (GstElement * element,
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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self->processed = 0;
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self->dropped = 0;
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self->expected_stream_time = GST_CLOCK_TIME_NONE;
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if (!gst_decklink_audio_src_open (self)) {
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ret = GST_STATE_CHANGE_FAILURE;
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goto out;
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@ -71,6 +71,11 @@ struct _GstDecklinkAudioSrc
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/* counter to keep track of timestamps */
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guint64 next_offset;
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/* detect gaps in stream time */
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GstClockTime expected_stream_time;
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guint64 processed;
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guint64 dropped;
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/* Last time we noticed a discont */
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GstClockTime discont_time;
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@ -36,6 +36,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_decklink_video_src_debug);
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#define DEFAULT_SKIP_FIRST_TIME (0)
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#define DEFAULT_DROP_NO_SIGNAL_FRAMES (FALSE)
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#ifndef ABSDIFF
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#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
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#endif
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enum
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{
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PROP_0,
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@ -825,6 +829,29 @@ gst_decklink_video_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
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}
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}
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/* 1 ns error can be just a rounding error, so that's OK. The Decklink
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* drivers give us a really steady stream time, so anything above 1 ns can't
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* be a rounding error and is therefore something to worry about */
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if (self->expected_stream_time != GST_CLOCK_TIME_NONE &&
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ABSDIFF (self->expected_stream_time, f.stream_timestamp) > 1) {
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GstMessage *msg;
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GstClockTime running_time;
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self->dropped += f.stream_timestamp - self->expected_stream_time;
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running_time = gst_segment_to_running_time (&GST_BASE_SRC (self)->segment,
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GST_FORMAT_TIME, f.timestamp);
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msg = gst_message_new_qos (GST_OBJECT (self), TRUE, running_time, f.stream_timestamp,
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f.timestamp, f.duration);
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gst_message_set_qos_stats (msg, GST_FORMAT_TIME, self->processed,
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self->dropped);
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gst_element_post_message (GST_ELEMENT (self), msg);
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}
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if (self->first_stream_time == GST_CLOCK_TIME_NONE)
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self->first_stream_time = f.stream_timestamp;
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self->processed = f.stream_timestamp - self->dropped - self->first_stream_time;
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self->expected_stream_time = f.stream_timestamp + f.stream_duration;
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g_mutex_unlock (&self->lock);
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if (caps_changed) {
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caps = gst_decklink_mode_get_caps (f.mode, f.format, TRUE);
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@ -1080,6 +1107,10 @@ gst_decklink_video_src_change_state (GstElement * element,
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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self->processed = 0;
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self->dropped = 0;
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self->expected_stream_time = GST_CLOCK_TIME_NONE;
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self->first_stream_time = GST_CLOCK_TIME_NONE;
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if (!gst_decklink_video_src_open (self)) {
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ret = GST_STATE_CHANGE_FAILURE;
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goto out;
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@ -58,6 +58,10 @@ struct _GstDecklinkVideoSrc
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gboolean output_stream_time;
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GstClockTime skip_first_time;
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gboolean drop_no_signal_frames;
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GstClockTime expected_stream_time;
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guint64 processed;
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guint64 dropped;
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guint64 first_stream_time;
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GstVideoInfo info;
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GstDecklinkVideoFormat video_format;
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