Sunday, 30 January 2011

Gstreamer Time & Sync

GstClock


Frist some introduce words:

absolute_time: the current time get from GstClock (monotonically increasing time)

running_time:the total time spent in the PLAYING state

base_time:defined as the absolute_time minus the running_time at the time when the pipeline is set to PLAYING

stream_time:the position in the stream(between 0 and the total duration)

GStreamer can use different clocks:


  • system time (with g_get_current_time() and with microsecond accuracy)
  • soundcards and other devices time(better, use get_clock in some elements to get a clock)
  • a network source based on packets received + timestamps in those packets (a typical example is an RTP source)

time is always expessed in nanoseconds, so I think there must have a conversion between different time types.

pipeline state and running_time:

  • NULL/READY, the running_time is undefined
  • PAUSED, the running_time remains at the time when it was last PAUSED
  • PLAYING, the running_time is the delta between the absolute_time and the base time
  • after a flushing seek, the running_time is set to 0(redistribute a new base_time to implement)

    running_time = absolute_time - base_time(In the PLAYING state)

GstBuffer

The GstBuffer timestamps and the preceeding NEW_SEGMENT event define a transformation of the buffer timestamps to running_time

B: GstBuffer

  • B.timestamp = buffer timestamp (GST_BUFFER_TIMESTAMP)

NS: NEWSEGMENT event preceeding the buffers

  • NS.start: start field in the NEWSEGMENT event
  • NS.stop: stop field in the NEWSEGMENT event
  • NS.rate: rate field of NEWSEGMENT event
  • NS.abs_rate: absolute value of rate field of NEWSEGMENT event(By default a pipeline will play from position 0 to
    the total duration of the media at a rate of 1.0)
  • NS.time: time field in the NEWSEGMENT event
  • NS.accum: total accumulated time of all previous NEWSEGMENT events. This field is kept in the GstSegment structure


    if (NS.rate > 0.0)
        B.running_time = (B.timestamp - NS.start) / NS.abs_rate + NS.accum
    else
        B.running_time = (NS.stop - B.timestamp) / NS.abs_rate + NS.accum
    

  • And we also got:

    stream_time = (B.timestamp - NS.start) * NS.abs_applied_rate + NS.time

  • This formula is typically used in sinks to report the current position in
    an accurate and efficient way:

    stream_time = (absolute_time - base_time - NS.accum) * NS.abs_rate * NS.abs_applied_rate + NS.time

Synchronisation

There are two ways to get the running_time:

  • using the clock and the element's base_time with:

    C.running_time = absolute_time - base_time

  • using the buffer timestamp and the preceeding NEWSEGMENT event as (assuming positive playback rate):

    B.running_time = (B.timestamp - NS.start) / NS.abs_rate + NS.accum

  • For synchronisation the following must hold:

    B.running_time = C.running_time

  • expaning:

    B.running_time = absolute_time - base_time

  • or:

    absolute_time = B.running_time + base_time

  • The absolute_time when a buffer with B.running_time should be played is noted with B.sync_time. Thus:

    B.sync_time = B.running_time + base_time

7 comments:

  1. Hello Sinkpad.
    Would you like to help us to solve a problem with a Gstreamer pipeline. Please see this thread:
    http://lists.freedesktop.org/archives/gstreamer-devel/2012-September/037233.html

    We have a GstCutter element that filtes out "silent" parts from a recording. This works perfectly well with MP3, FLAC and WAV encoders.

    But some encoder (AAC, OGG Vorbis and SPX) add silent parts back to the media when replayed. This happens because the buffers have an internal timestamp that fills the "silent" gaps.

    I have been asked to push and a "new segment event" to adjust the timestamps, but I haven't succeeded with this. I do not have enough knowledge about Gstreamer. Please help me, help us? We will use this to improve our free and GPL'ed audio-recorder (many people use it already). See: https://launchpad.net/audio-recorder

    Notice also: I have actually replaced GstCutter with GstVader (a VAD, voice detection filter) element. It performs better than the standard GstCutter element. Please read the thread. Most kindly Alexander.


    ReplyDelete
  2. Very good article. I'm dealing with some of these issues as well..
    My site best renovation in orlando fl

    ReplyDelete
  3. Hello, Neat post. There's an issue together with your website in web explorer, might test this? IE nonetheless is the market chief and a huge component to other people will omit your excellent writing due to this problem.

    Check out my web site; fingerprint building in thailand
    Also see my page: Executive dirtbag

    ReplyDelete
  4. It's an awesome paragraph designed for all the internet visitors; they will take benefit from it I am sure.

    Here is my web blog; http://xxx-fuck.net

    ReplyDelete
  5. Somebody essentially lend a hand to make severely posts I'd state. That is the very first time I frequented your web page and thus far? I amazed with the analysis you made to create this particular publish extraordinary. Excellent process!

    Feel free to visit my blog post; www.cuteteenporn.net

    ReplyDelete
  6. Cool and I have a super provide: Where To Remodel House bungalow exterior makeover

    ReplyDelete