Synchronised RTP Client

Mon May 30 09:08:00 UTC 2016

Doh! Looks like I might have missed this:

In my factory subclass:

self.set_media_gtype(MyRTSPMediaType)

Which should reference:

class MyRTSPMediaType(GstRtspServer.RTSPMedia):
   def do_setup_rtpbin(self, rtpbin):
      rtpbin.set_property('ntp-time-source', 'clock-time')
      return(rtpbin)

And suddenly my timestamp issues seem to go away :)

On 29 May 2016 at 23:07, Jonathan Thorpe <jt at jonthorpe.net> wrote:

> Hi All,
>
> Sebastian - thank you so much for pointing me in the right direction with
> this earlier. It was very helpful in allowing me to progress with this and
> I have finally had a chance to put some code together.
>
> I have written an RTSP server in Python trying to re-implement the code
> that was in test-netclock.c / test-netclock-client.c, however I have a few
> problems with synchronisation. I must be close.
>
> My server is configured with a pipeline as follows:
>
> My launch string for the factory on the server is as follows:
>
> udpsrc uri=udp://127.0.0.1:6001 caps="application/x-rtp,
> media=(string)audio, clock-rate=(int)44100, channels=(int)2,
> format=(string)S16LE" ! rtpL16depay ! audioconvert ! audioresample !
> opusenc bitrate=96000 inband-fec=1 ! rtpopuspay name=pay0
>
> It captures raw audio through RTP on port 6001 and converts it to OPUS.
> which will be sent out to the clients (which should all play out in sync).
>
> For this factory, I'm also setting:
>
> factory.set_shared(True)
> factory.set_retransmission_time(5000)
> factory.set_clock(clock)
> factory.set_profiles(GstRtsp.RTSPProfile.AVPF)
> Where clock is a time provider that I want to run from the server and use
> that to sync the clients:
>
> clock = Gst.SystemClock.obtain()
> clock_provider = GstNet.NetTimeProvider.new(clock, None, 8555)
>
> On my clients, where I have sync=true on my alsaaudiosink, I get about 2
> seconds of audio followed by silence and a lot of this:
>
> audiobasesink
> gstaudiobasesink.c:1807:gst_audio_base_sink_get_alignment:<autoaudiosink0-actual-sink-alsa>
> Unexpected discontinuity in audio timestamps of +1012391:15:11.969041666,
> resyncing
>
> audiobasesink
> gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<autoaudiosink0-actual-sink-alsa>
> correct clock skew -0:00:00.020016938 < -+0:00:00.020000000
>
> The client comprises the following pipeline (location is the server and
> buffer is usually configured as 1000 or 5000)
>
> My client contains the following:
>
> self.pipeline = Gst.parse_launch('rtspsrc location=%s latency=%s
> buffer-mode=synced ntp-time-source=clock-time ntp-sync=1 do-rtcp=true !
> rtpopusdepay name=pay0 ! opusdec ! audioconvert ! autoaudiosink sync=true
> async=false' % (src, latency))
>
> # make a clock slaving to the network
> self.clock = GstNet.NetClientClock.new('clock0', clock_ip, clock_port, 0)
>
> # Wait for initial clock sync
> self.clock.wait_for_sync(Gst.CLOCK_TIME_NONE)
>
> self.pipeline.use_clock(self.clock)
> self.pipeline.set_start_time(Gst.CLOCK_TIME_NONE)
> self.pipeline.set_latency(latency * Gst.MSECOND)
>
> self.pipeline.set_state(Gst.State.PLAYING)
>
> # Wait until error or EOS.
> self.bus = self.pipeline.get_bus()
> self.bus.add_signal_watch()
> self.bus.connect('message', self.bus_handler)
>
> The intention for this set up is to:
> 1. Buffer 1-5 seconds of audio.
> 2. Within this buffer period, allow for retransmits.
> 3. Otherwise, keep the clients in sync with the server's clock using the
> GST NetClientClock.
> This configuration plays audio fine if I change "sync=false" on the audio
> sink, but this also breaks synchronisation.
>
> Any pointers as to what I've done wrong? I'm currently running GStreamer
> out of git.
>
> Kind Regards,
> Jonathan
>
>
>
>
>> Message: 5
>> Date: Mon, 25 Apr 2016 09:24:53 +0300
>> From: Sebastian Dröge <sebastian at centricular.com>
>> To: Discussion of the development of and with GStreamer
>>         <gstreamer-devel at lists.freedesktop.org>
>> Subject: Re: Synchronised RTP Clients
>> Message-ID: <1461565493.2297.107.camel at centricular.com>
>> Content-Type: text/plain; charset="utf-8"
>>
>> On Mo, 2016-04-25 at 13:19 +1000, Jonathan Thorpe wrote:
>> >
>> > I have clearly chosen a very complex project to get started with
>> > GStreamer on, but would be very grateful if someone could assist with
>> > helping establish pipelines (Client and Server) that will achieve
>> > this.
>>
>> Take a look at my presentation from last year's GStreamer conference:
>>
>> https://gstconf.ubicast.tv/videos/synchronised-multi-room-media-playback-and-distributed-live-media-processing-and-mixing-with-gstreamer/
>>
>> Also these two example applications are basically doing exactly what
>> you want, just with the GStreamer netclock instead of PTP. But
>> exchanging that is a matter of changing a couple of lines.
>>
>>
>> https://cgit.freedesktop.org/gstreamer/gst-rtsp-server/tree/examples/test-netclock.c
>>
>> https://cgit.freedesktop.org/gstreamer/gst-rtsp-server/tree/examples/test-netclock-client.c
>>
>> The examples are using RTSP and gst-rtsp-server, but there's nothing
>> RTSP specific in how all this works. It's just for simplicity to make
>> it easier to exchange the SDP between sender and clients.
>>
>> --
>> Sebastian Dröge, Centricular Ltd · http://www.centricular.com
>>
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>> ------------------------------
>>
>> Message: 6
>> Date: Mon, 25 Apr 2016 09:20:48 +0100
>> From: Andy Robinson <andy at seventhstring.com>
>> To: gstreamer-devel at lists.freedesktop.org
>> Subject: Re: How to query the GstSegment "rate"?
>> Message-ID: <571DD360.10109 at seventhstring.com>
>> Content-Type: text/plain; charset=utf-8; format=flowed
>>
>> On 25/04/16 07:21, Sebastian Dröge wrote:
>> > On Sa, 2016-04-23 at 14:55 +0100, Andy Robinson wrote:
>> >> I imagine there must be some way of querying the
>> >> pipeline for the rate once it is in paused state, but how?
>> >
>> > You could try the SEGMENT query, if something in your pipeline is
>> > answering it then it will contain the rate.
>> >
>> > But the bigger question is why you need to query the rate and don't
>> > know it already. In the end it was your code that was setting that
>> > exact rate via a seek :)
>> >
>> >
>> > Also what's the bigger picture here? Why do you want to convert from
>> > stream time to the scaled stream time by rate (which is not exactly the
>> > running time in general, but maybe you actually want the running
>> > time?)?
>> >
>>
>> Even if I don't do any seek, a Segment event goes down the pipeline and
>> it has a rate of 0.5 - I know this from putting diagnostics in the
>> pipeline.
>>
>> The file is here if you are interested:
>> http://www.seventhstring.com/other2/JAttendraiShort50.mov
>> It plays ok in Parole Media Player on Linux, or QuickTime on Mac. It's
>> 60 secs long and was produced, using QuickTime, by slowing down a 30 sec
>> clip to half speed.
>>
>> When I play it in my app, I find that when I want to seek using
>> gst_element_seek_simple within this video I must use the pre-slowdown
>> times, e.g. if I want to seek to the 40th second of the 60 second video,
>> I must actually seek to a time of 20 secs, and also must make a similar
>> adjustment to the values returned by gst_element_query_position.
>>
>> It seems to me that I need to get that 0.5 rate number and use it as a
>> multiplier when I call gst_element_seek_simple but if there is a better
>> way, please advise me.
>>
>> Of course, maybe the file is simply erroneous, illegal, though it does
>> play ok in some (not all) players.
>>
>> Regards,
>> Andy Robinson, Seventh String Software, www.seventhstring.com
>>
>>
>> ------------------------------
>>
>> Message: 7
>> Date: Mon, 25 Apr 2016 08:28:23 +0200
>> From: marc dingemans <marcdingemans62 at gmail.com>
>> To: gstreamer-devel at lists.freedesktop.org
>> Subject: muliple gst_element_seek for creating new file
>> Message-ID:
>>         <CAN2V=
>> fCEOhsBCx8YPfWdiL5OXcdYHQEcqP1cED0vO_V49SdC9A at mail.gmail.com>
>> Content-Type: text/plain; charset="utf-8"
>>
>> Hi
>>
>> I want to extract some video parts from a mkv file and create based on
>> these parts one new mkv file.
>>
>> This is done with muliple gst_element_seek calls with a time frame.
>> When a eos is received, then a the next gst_element_seek is started until
>> done.
>> The destination file is generated, but can not be played with VLC.
>> The seek is done executed on element "matroskademux" (also used pipeline)
>> with GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_SNAP_BEFORE properties
>>
>> Used pipeline :
>> filesrc| matroskademux| h264parse | matroskamux | filesink
>>
>>
>> When I use next pipeline, then everything works fine.
>> filesrc| matroskademux| h264parse | avdec_h264| autovideosink
>>
>>
>> Why does the seek seems work to gui and not to file.
>> Must I use other approaches to put parts from a file in a new one
>>
>> Thanks for your help
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