tsdemux/tsparse TODO -------------------- * clock for live streams In order for playback to happen at the same rate as on the producer, we need to estimate the remote clock based on capture time and PCR values. For this estimation to be as accurate as possible, the capture time needs to happen on the sources. => Ensure live sources actually timestamp their buffers Once we have accurate timestamps, we can use an algorithm to calculate the PCR/local-clock skew. => Use the EPTLA algorithm as used in -good/rtp/rtpmanager/ gstrtpjitterbuffer * Seeking => Split out in a separate file/object. It is polluting tsdemux for code readability/clarity. * Perfomance : Creation/Destruction of buffers is slow * => This is due to g_type_instance_create using a dogslow rwlock which take up to 50% of gst_adapter_take_buffer() => Bugzilla #585375 (performance and contention problems) * mpegtspacketizer * offset/timestamp of incoming buffers need to be carried on to the sub-buffers in order for several demuxer features to work correctly. * mpegtsparser * SERIOUS room for improvement performance-wise (see callgrind) 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. 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. => A delta, DTS_delta between incoming buffer timestamp and DTS/PTS needs to be computed. => The outgoing buffers will be timestamped with their PTS values (overflow corrected) offseted by that initial DTS_delta. 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. It corresponds to the biggest Access Unit spacing, in this case 1/video-framerate. 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 demuxer will therefore report an added latency of 1s to handle this interleave. 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. => A delta between the first DTS to output and the segment start position needs to be computed. => The outgoing buffers will be timestamped with their PTS values (overflow corrected) offseted by that initial delta. 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 base stream/running time corresponds to the DTS of the first buffer we will output. The DTS_delta becomes that earliest DTS. => FILLME X) General notes It is assumed that PTS/DTS rollovers are detected and corrected such as the outgoing timestamps never rollover. This can be easily handled by correcting the DTS_delta when such rollovers are detected. The maximum value of a GstClockTimeDiff is almost 3 centuries, we therefore have enough margin to handle a decent number of rollovers. The generic equation for calculating outgoing buffer timestamps therefore becomes: D = DTS_delta, with rollover corrections PTS = PTS of the buffer we are going to push out TS = Timestamp of the outgoing buffer ==> TS = PTS + D If seeking is handled upstream for push-based cases, whether live or not, no extra modification is required. If seeking is handled by the demuxer in the non-live push-based cases (converting from TIME to BYTES), the demuxer will need to set the segment start/time values to the requested seek position. The DTS_delta will also have to be recomputed to take into account the seek position. [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.