1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
|
tsdemux/tsparse TODO
--------------------
* Perfomance
* Bufferlist : Creating/Destroying very small buffers is too
costly. Switch to pre-/re-allocating outgoing buffers in which we
copy the data.
* Adapter : Use gst_adapter_peek()/_flush() instead of constantly
creating buffers.
* Latency
* Calculate the actual latency instead of returning a fixed
value. The latency (for live streams) is the difference between the
currently inputted buffer timestamp (can be stored in the
packetizer) and the buffer we're pushing out.
This value should be reported/updated (leave a bit of extra margin
in addition to the calculated value).
* mpegtsparser
* SERIOUS room for improvement performance-wise (see callgrind),
mostly related to performance issues mentionned above.
* Random-access seeking
* Do minimal parsing of video headers to detect keyframes and use
that to compute the keyframe intervals. Use that interval to offset
the seek position in order to maximize the chance of pushing out the
requested frames.
Synchronization, Scheduling and Timestamping
--------------------------------------------
A mpeg-ts demuxer can be used in a variety of situations:
* lives streaming over DVB, UDP, RTP,..
* play-as-you-download like HTTP Live Streaming or UPNP/DLNA
* random-access local playback, file, Bluray, ...
Those use-cases can be categorized in 3 different categories:
* Push-based scheduling with live sources [0]
* Push-based scheduling with non-live sources
* Pull-based scheduling with fast random-access
Due to the nature of timing within the mpeg-ts format, we need to
pay extra attention to the outgoing NEWSEGMENT event and buffer
timestamps in order to guarantee proper playback and synchronization
of the stream.
In the following, 'timestamps' correspond to GStreamer
buffer/segment values. The mpeg-ts PCR/DTS/PTS values are indicated
with their actual name.
1) Live push-based scheduling
The NEWSEGMENT event will be in time format and is forwarded as is,
and the values are cached locally.
Since the clock is running when the upstream buffers are captured,
the outgoing buffer timestamps need to correspond to the incoming
buffer timestamp values.
=> mpegtspacketizer keeps track of PCR and input timestamp and
extrapolates a clock skew using the EPTLA algorithm.
=> The outgoing buffers will be timestamped with their PTS values
(overflow corrected) corrected by that calculated clock skew.
A latency is introduced between the time the buffer containing the
first bit of a Access Unit is received in the demuxer and the moment
the demuxer pushed out the buffer corresponding to that Access Unit.
=> That latency needs to be reported.
According to the ISO/IEC 13818-1:2007 specifications, D.0.1 Timing
mode, the "coded audio and video that represent sound and pictures
that are to be presented simultaneously may be separated in time
within the coded bit stream by ==>as much as one second<=="
=> The algorithm to calculate the latency should take that into
account.
2) Non-live push-based scheduling
If the upstream NEWSEGMENT is in time format, the NEWSEGMENT event
is forwarded as is, and the values are cached locally.
If upstream does provide a NEWSEGMENT in another format, we need to
compute one by taking the default values:
start : 0
stop : GST_CLOCK_TIME_NONE
time : 0
Since no prerolling is happening downstream and the incoming buffers
do not have capture timestamps, we need to ensure the first buffer
we push out corresponds to the base segment start runing time.
=> The packetizer keeps track of PCR locations and offsets in
addition to the clock skew (in the case of upstream buffers
being timestamped, which is the case for HLS).
=> The demuxer indicates to the packetizer when he sees the
'beginning' of the program (i.e. the first valid PAT/PMT
combination). The packetizer will then use that location as
"timestamp 0", or "reference position/PCR".
=> The lowest DTS is passed to the packetizer to be converted to
timestamp. That value is computed in the same way as live
streams if upstream buffers have timestamps, or will be
subtracted from the reference PCR.
=> The outgoing buffers will be timestamped with their PTS values
(overflow corrected) adjusted by the packetizer.
Latency is reported just as with the live use-case.
3) Random access pull-mode
We do not get a NEWSEGMENT event from upstream, we therefore need to
compute the outgoing values.
=> The outgoing values for the newsegment are calculated like for
the non-live push-based mode when upstream doesn't provide
timestamp'ed buffers.
=> The outgoing buffer timestamps are timestamped with their PTS
values (overflow corrected) adjusted by the packetizer.
[0] When talking about live sources, we mean this in the GStreamer
definition of live sources, which is to say sources where if we miss
the capture, we will miss the data to be captured. Sources which do
internal buffering (like TCP connections or file descriptors) are
*NOT* live sources.
|
