VP8 - Dropping inter-frame before intra-frame
bojan74
bojan.flander at gmail.com
Thu Mar 23 18:06:52 UTC 2017
Hello,
I have a pipeline like this:
appsrc->rtpjitterbuffer->rtpvp8depay->webmmux->filesink
appsrc use push-buffer to push RTP header+payload to pipe line.
I have some old VP8 pcap example that I am using as source - I extract RTP
header+payload and use it as input.
And it works well, I get .webm file that I can play.
But VP8 .pcap that I created now and use it as source in the same way as
above produce me this error:
0:00:02.287498382 10328 0000014BEC2044C0 INFO rtpvp8depay
gstrtpvp8depay.c:197:gst_rtp_vp8_depay_process:<rtpvp8depay0> *Dropping
inter-frame before intra-frame*
Every packet is dropped and so webm file is without content.
In both cases I use same code (application), just input file (RTP packets
passing to appsrc) is different.
I compare both GST_DEBUG outputs.
When pipeline is change to PLAYING I get in both cases:
0:00:02.146507677 12124 0000019EC3594000 INFO GST_STATES
gstelement.c:2419:gst_element_continue_state:<rtpjitterbuffer0> completed
state change to PLAYING
0:00:02.146567270 12124 0000019EC3BF4540 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps:* creating caps event application/x-rtp,
media=(string)video, clock-rate=(int)90000, encoding-name=(string)VP8*
0:00:02.146720225 12124 0000019EC3594000 INFO GST_STATES
gstelement.c:2324:_priv_gst_element_state_changed:<rtpjitterbuffer0>
notifying about state-changed PAUSED to PLAYING (VOID_PENDING pending)
On 1st input RTP packet these lines are very similar in both cases (working,
non-working):
0:00:02.181052985 12124 0000019EC3BF4540 INFO GST_EVENT
gstevent.c:889:gst_event_new_segment: creating segment event time segment
start=0:00:00.000000000, offset=0:00:00.000000000, stop=99:99:99.999999999,
rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000,
base=0:00:00.000000000, position 0:00:00.000000000, duration
99:99:99.999999999
0:00:02.183867423 12124 0000019EC3BF4540 INFO basesrc
gstbasesrc.c:2847:gst_base_src_loop:<appsrc0> marking pending DISCONT
0:00:02.184271661 12124 0000019EC3BF4540 INFO rtpjitterbuffer
rtpjitterbuffer.c:855:rtp_jitter_buffer_insert: resync to time
0:00:00.000000000, rtptime 11:34:16.601977777
0:00:02.184518640 12124 0000019EC3BF44C0 INFO rtpjitterbuffer
gstrtpjitterbuffer.c:3788:do_deadline_timeout:<rtpjitterbuffer0> got
deadline timeout
0:00:02.184999025 12124 0000019EC3BF4500 INFO GST_EVENT
gstevent.c:889:gst_event_new_segment: creating segment event time segment
start=0:00:00.000000000, offset=0:00:00.000000000, stop=99:99:99.999999999,
rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000,
base=0:00:00.000000000, position 0:00:00.000000000, duration
99:99:99.999999999
But difference is on 2nd packet where I got these lines only with working
example:
0:00:02.260681603 12124 0000019EC3BF4500 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: *creating caps event video/x-vp8,
framerate=(fraction)0/1, height=(int)480, width=(int)640, profile=(string)0*
0:00:02.263667204 12124 0000019EC3BF4500 FIXME basesink
gstbasesink.c:3077:gst_base_sink_default_event:<filesink0> stream-start
event without group-id. Consider implementing group-id handling in the
upstream elements
0:00:02.263879421 12124 0000019EC3BF4500 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: creating caps event video/webm
0:00:02.264077070 12124 0000019EC3BF4500 INFO matroskamux
matroska-mux.c:2765:gst_matroska_mux_start:<ebmlwrite0> DocType: webm,
Version: 2
0:00:02.264275713 12124 0000019EC3BF4500 INFO ebmlwrite
ebml-write.c:216:gst_ebml_writer_send_segment_event: seeking to 0
0:00:02.264457471 12124 0000019EC3BF4500 INFO GST_EVENT
gstevent.c:889:gst_event_new_segment: creating segment event time segment
start=0:00:00.000000000, offset=0:00:00.000000000, stop=99:99:99.999999999,
rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000,
base=0:00:00.000000000, position 0:00:00.000000000, duration
99:99:99.999999999
0:00:02.265983048 12124 0000019EC3BF4500 INFO GST_EVENT
gstevent.c:808:gst_event_new_caps: creating caps event video/webm,
streamheader=(buffer)<
1a45dfa301000000000000104282857765626d0042878102428581021853806701ffffffffffffff1549a96601000000000000562ad7b1830f42404d80a54753747265616d6572206d6174726f736b616d75782076657273696f6e20312e31302e32005741994753747265616d6572204d6174726f736b61206d7578657200446188071aea7b4fc28e001654ae6b010000000000003cae0100000000000033d7810183810173c588c328bb32a40e3376536e86566964656f00e00100000000000008b0820280ba8201e08686565f56503800
>
Any idea why this last "creating caps" doesnt happen in non-working case?
Is some info missing in RTP stream?
Or do I have to set any additional caps for this case?
Or GST_BUFFER_PTS or GST_BUFFER_DURATION are not calculated well before I
push-buffer?
...
Thanks a lot in advance for any hint!
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