'Re: Unsynced Audio and Video signals in RTSP Stream'

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List:       gstreamer-devel
Subject:    Re: Unsynced Audio and Video signals in RTSP Stream
From:       vinod kesti via gstreamer-devel <gstreamer-devel () lists ! freedesktop ! org>
Date:       2024-04-23 15:46:31
Message-ID: 1735400834.1779776.1713887191343 () mail ! yahoo ! com
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Try sending SR report. The NTP timestamp in the SR report will help in audio and video \
synchronization.Check rtpbin examples which have SR and RR reort


Sent from Yahoo Mail. Get the app 

    On Monday, 22 April, 2024 at 03:20:51 pm GMT-5, cfd new via gstreamer-devel \
<gstreamer-devel@lists.freedesktop.org> wrote:    
  I guess you send out audio and video separately. If yes, you may try to send out audio first \
and then video.  Joe

    On Thursday, April 18, 2024, 08:13:43 a.m. EDT, Felix Sternkopf via gstreamer-devel \
<gstreamer-devel@lists.freedesktop.org> wrote:    
  <!--#yiv4638044512 filtered {}#yiv4638044512 filtered {}#yiv4638044512 filtered \
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sans-serif;}#yiv4638044512 filtered {}#yiv4638044512 div.yiv4638044512WordSection1 {}--> Dear \
GST Community,  
i have implemented a rtsp server with the gst-rtsp-server module. The server receives an audio \
signal via an interaudiosrc and a video signal via an intervideosrc:  
                      intervideosrc channel=videosrc Ă  videoconvertĂ  vaapih264enc \
bitrate=5000 zerolatency=trueĂ  rtph264pay pt=96 name=pay0  
                       interaudiosrc channel=audiosrcĂ  opusenc bitrate=64Ă  queueĂ  rtpopuspay \
pt=97 name=pay1  
   
 
When I try to view the stream via an rtsp client, for example VLC Player the stream works, but \
the audio signal is about 1 second behind the video signal. I already tried different audio \
encoders and different queue parameters like buffer size and leaky queues, but nothing solved \
the problem.  
When I start the same pipeline as above but mux both signals and write them into a filesink \
they are synchronized.  
   
 
I know that the server module internally uses the multiudpsink element. Is there any \
possibility to edit multiudpsink parameters from outside?  
Does anybody have any ideas about this problem? I already read every google result I could find \
about it.  
   
 
Best Regards,
 
Felix Sternkopf
     


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<html><head></head><body><div class="ydpb1b69bc8yahoo-style-wrap" style="font-family:Helvetica \
Neue, Helvetica, Arial, sans-serif;font-size:13px;"><div><div dir="ltr" data-setdir="false">Try \
sending SR report. The NTP timestamp in the SR report will help in audio and video \
synchronization.</div><div dir="ltr" data-setdir="false">Check rtpbin examples which have SR \
and RR reort</div><div><br></div><div \
class="ydpb1b69bc8signature"><div><br></div><div><br></div><div>Sent from Yahoo Mail. <a \
href="https://yho.com/148vdq" rel="nofollow" target="_blank">Get the app</a></div></div></div>  \
<div><br></div><div><br></div>  
        </div><div id="yahoo_quoted_4535870060" class="yahoo_quoted">
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sans-serif;font-size:13px;color:#26282a;">  
                <div>
                        On Monday, 22 April, 2024 at 03:20:51 pm GMT-5, cfd new via \
gstreamer-devel &lt;gstreamer-devel@lists.freedesktop.org&gt; wrote:  </div>
                    <div><br></div>
                    <div><br></div>
                
                
                <div><div id="yiv4638044512"><div><div style="font-family:Helvetica Neue, \
Helvetica, Arial, sans-serif;font-size:13px;" \
class="yiv4638044512ydp78cdc2d0yahoo-style-wrap"><div></div>  <div dir="ltr">I guess you send \
out audio and video separately. If yes, you may try to send out audio first and then \
video.</div><div dir="ltr"><br clear="none"></div><div dir="ltr">&nbsp;&nbsp; Joe<br \
clear="none"></div><div><br clear="none"></div>  
        </div><div id="yiv4638044512yqt63953" class="yiv4638044512yqt9414871302"><div \
                id="yiv4638044512yahoo_quoted_4094826741" class="yiv4638044512yahoo_quoted">
            <div style="font-family:'Helvetica Neue', Helvetica, Arial, \
sans-serif;font-size:13px;color:#26282a;">  
                <div>
                        On Thursday, April 18, 2024, 08:13:43 a.m. EDT, Felix Sternkopf via \
gstreamer-devel &lt;gstreamer-devel@lists.freedesktop.org&gt; wrote:  </div>
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<p class="yiv4638044512MsoNormal"><span style="font-size:10.0pt;font-family:sans-serif;">Dear \
GST Community,</span></p>  <p class="yiv4638044512MsoNormal"><span \
style="font-size:10.0pt;font-family:sans-serif;">i have implemented a rtsp server with the \
gst-rtsp-server module. The server receives an audio signal via an interaudiosrc and a video \
signal via an intervideosrc:</span></p>  <p class="yiv4638044512MsoNormal"><span \
style="font-size:10.0pt;font-family:sans-serif;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
 <i>intervideosrc channel=videosrc </i></span><i><span \
style="font-size:10.0pt;font-family:Wingdings;">Ă </span></i><i><span \
style="font-size:10.0pt;font-family:sans-serif;"> videoconvert </span></i><i><span \
style="font-size:10.0pt;font-family:Wingdings;">Ă </span></i><i><span \
style="font-size:10.0pt;font-family:sans-serif;"> vaapih264enc bitrate=5000 zerolatency=true \
</span></i><i><span style="font-size:10.0pt;font-family:Wingdings;">Ă </span></i><i><span \
style="font-size:10.0pt;font-family:sans-serif;"> rtph264pay pt=96 name=pay0</span></i></p>  <p \
class="yiv4638044512MsoNormal"><i><span \
style="font-size:10.0pt;font-family:sans-serif;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; \
interaudiosrc channel=audiosrc </span></i><i><span \
style="font-size:10.0pt;font-family:Wingdings;">Ă </span></i><i><span \
style="font-size:10.0pt;font-family:sans-serif;"> opusenc bitrate=64 </span></i><i><span \
style="font-size:10.0pt;font-family:Wingdings;">Ă </span></i><i><span \
style="font-size:10.0pt;font-family:sans-serif;"> queue </span></i><i><span \
style="font-size:10.0pt;font-family:Wingdings;">Ă </span></i><i><span \
style="font-size:10.0pt;font-family:sans-serif;"> rtpopuspay pt=97 name=pay1</span></i></p>  <p \
class="yiv4638044512MsoNormal"><span style="font-size:10.0pt;font-family:sans-serif;"> \
&nbsp;</span></p>  <p class="yiv4638044512MsoNormal"><span \
style="font-size:10.0pt;font-family:sans-serif;">When I try to view the stream via an rtsp \
client, for example VLC Player the stream works, but the audio signal is about 1 second behind \
the video signal. I already tried different  audio encoders and different queue parameters like \
buffer size and leaky queues, but nothing solved the problem.</span></p>  <p \
class="yiv4638044512MsoNormal"><span style="font-size:10.0pt;font-family:sans-serif;">When I \
start the same pipeline as above but mux both signals and write them into a filesink they are \
synchronized.</span></p>  <p class="yiv4638044512MsoNormal"><span \
style="font-size:10.0pt;font-family:sans-serif;"> &nbsp;</span></p>  <p \
class="yiv4638044512MsoNormal"><span style="font-size:10.0pt;font-family:sans-serif;">I know \
that the server module internally uses the multiudpsink element. Is there any possibility to \
edit multiudpsink parameters from outside?</span></p>  <p class="yiv4638044512MsoNormal"><span \
style="font-size:10.0pt;font-family:sans-serif;">Does anybody have any ideas about this \
problem? I already read every google result I could find about it.</span></p>  <p \
class="yiv4638044512MsoNormal"><span style="font-size:10.0pt;font-family:sans-serif;"> \
&nbsp;</span></p>  <p class="yiv4638044512MsoNormal"><span \
style="font-size:10.0pt;font-family:sans-serif;">Best Regards,</span></p>  <p \
class="yiv4638044512MsoNormal"><span style="font-size:10.0pt;font-family:sans-serif;">Felix \
Sternkopf</span></p>  </div>
</div>

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