[Asterisk-video] MP4Play async Audio / Video

Klaus Darilion klaus.mailinglists at pernau.at
Tue Sep 25 10:09:21 CDT 2007


Which parameters have you used for encoding with ffmpeg?

Thomas Z. schrieb:
> Hi Sergio,
> 
> thank you very much. The result with the Helix Mobile Producer is much
> better.
> But we need a software runs on a Linux environment.
> Can you support us with a another linux software?
> 
> Regards
> Thomas
> 
> 
> Sergio Garcia schrieb:
>> You coulde try with qt encoder or the helix mobile producer.. 
>>
>> ---------- Original Message ----------------------------------
>> From: "Thomas Z." <mobilemail at gmx-topmail.de>
>> Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
>> Date:  Tue, 25 Sep 2007 10:52:07 +0200
>>
>>   
>>> Hello Sergio,
>>>
>>> do you know a freeware or cheap CBR encoder?
>>>
>>> Regards
>>> Thomas
>>>
>>> Sergio Garcia Murillo schrieb:
>>>     
>>>> Hi Thomas, Thomas and Ratmin.. :)
>>>>  
>>>> Back to work form a long abroad weekend (nice Belgium!)
>>>>  
>>>> As Ratmin has stated the Skew indication would help to indicate the
>>>> handset the delay between audio and video.
>>>> It's usually used to correct fixed delays introduced by internal
>>>> jitter buffers and thinks alike. The problem that Thomas
>>>> Z. is suffering seems to be produced by using ffmpeg for video
>>>> encoding (as Thomas without Z points :)
>>>> The problem is the following, ffmpeg is a great VBR (variable bit
>>>> rate) encoder, but I've not been able to behave like a
>>>> CBR (constant bit rate) encoder. The problem is that it's prepared for
>>>> movies, not for streaming, and defenitevilly not
>>>> at this low bitrates.
>>>> The inner loop of the encoder always tries to encode ALL macroblocks
>>>> in a frame, so in I frames and at a very low
>>>> bitrate it simply just can't get a small amount of data for that frame
>>>> and when it's enqued into h324m it takes more
>>>> than one frame time to send (sometimes even more than a second), the
>>>> following frames are small and sent after
>>>> that until the queue is empty, which causes a fast-forward effect
>>>> on some handsets. As you can imagine adjustime
>>>> audio to that scenary is jus impossible (even with the Skew Indication
>>>> the result would be horrible!!)
>>>>  
>>>> The good solution: use a CBR encoder :)
>>>> The not so good solution: drop packets when the queue is filled, but
>>>> you'll end up dropping all the frames between
>>>> each I frame.. but in that case it would be much better to lower the
>>>> video fps...
>>>>  
>>>> BR
>>>> Sergio
>>>>
>>>>     ----- Original Message -----
>>>>     *From:* Ramtin Amin <mailto:keytwho at hotmail.com>
>>>>     *To:* Development discussion of video media support in Asterisk
>>>>     <mailto:asterisk-video at lists.digium.com>
>>>>     *Sent:* Monday, September 24, 2007 2:01 PM
>>>>     *Subject:* Re: [Asterisk-video] MP4Play async Audio / Video
>>>>
>>>>
>>>>
>>>>
>>>>     RTSP has timestamp... so you can usually know how to sync the
>>>>     audio/video
>>>>     But the problem would be more that we'll need to add a Skew
>>>>     Indication concept inside asterisk's core so it could make the
>>>>     message go throught the different channel/application, as it is
>>>>     currently done for VideoFastUpdatePicutre... Which means that
>>>>     currently, if a SIP channel receives a VideoFastUpdatePicture, it
>>>>     is capable of resending it to an other channel...
>>>>
>>>>      
>>>>     ------------------------------------------------------------------------
>>>>
>>>>     > Date: Mon, 24 Sep 2007 13:48:54 +0200
>>>>     > From: mobilemail at gmx-topmail.de
>>>>     > To: asterisk-video at lists.digium.com
>>>>     > Subject: Re: [Asterisk-video] MP4Play async Audio / Video
>>>>     >
>>>>     > Hello Ramin,
>>>>     > hello Thomas.
>>>>     >
>>>>     > I think both ways are possible. But I don't know if it easy to
>>>>     implement.
>>>>     > Sergio, what do you think about it?
>>>>     >
>>>>     > Is this also a problem if we stream the audi/video via rtsp?
>>>>     >
>>>>     > Regrads
>>>>     > Thomas
>>>>     >
>>>>     >
>>>>     > Ramtin Amin schrieb:
>>>>     > > hello
>>>>     > >
>>>>     > > The way I solved this problem was by sending a H223SkewIndication
>>>>     > > message to the other terminal
>>>>     > > Acutally, You will have to ask Sergio to add a H245 indication
>>>>     message
>>>>     > > of type h223SkewIndication and when playing the video with
>>>>     mp4play, he
>>>>     > > will have to see the difference of timing between Video and
>>>>     Audio and
>>>>     > > then send this value to the remote terminal so the lip sync
>>>>     would work...
>>>>     > >
>>>>     > >
>>>>     > >
>>>>     > > **
>>>>     > >
>>>>     > > *2.3 Multipoint Lip Synchronization*
>>>>     > >
>>>>     > > In a multipoint VC, each terminal may transmit different
>>>>     > >
>>>>     > > //
>>>>     > >
>>>>     > > /H223SkewIndication /message for associated video and
>>>>     > >
>>>>     > > audio channels in H.223 protocol. To enable lip
>>>>     > >
>>>>     > > synchronization at receiving terminals, MCUs will
>>>>     > >
>>>>     > > transmit accurate /H223SkewIndication /messages. MCUs
>>>>     > >
>>>>     > > may accomplish this by adding delay to equalize the
>>>>     > >
>>>>     > > audio/video skew for all transmitting terminals. When
>>>>     > >
>>>>     > > switching between broadcasting terminals, H.223 may
>>>>     > >
>>>>     > > transmit a new /H223SkewIndication /message reflecting the
>>>>     > >
>>>>     > > audio/video skew of the current broadcaster.
>>>>     > >
>>>>     > >
>>>>     > > --
>>>>     > > Ramtin Amin
>>>>     > >
>>>>     > >
>>>>     > >
>>>>     > >
>>>>     > >
>>>>     ------------------------------------------------------------------------
>>>>     > >
>>>>     > > > From: thomas.frieling at viif.de
>>>>     > > > To: asterisk-video at lists.digium.com
>>>>     > > > Date: Mon, 24 Sep 2007 11:48:12 +0200
>>>>     > > > Subject: Re: [Asterisk-video] MP4Play async Audio / Video
>>>>     > > >
>>>>     > > > Hi Thomas Z!
>>>>     > > >
>>>>     > > > I think this problem is due to the bitrate restriction on 3G
>>>>     calls. The
>>>>     > > > audio stream is always sent immediately while the video
>>>>     stream has to
>>>>     > > > use what is left of the bandwidth. This is why videos become
>>>>     synchrous
>>>>     > > > again when the video bitrate is pretty low for a while in
>>>>     the video...
>>>>     > > >
>>>>     > > > Take a look at this discussion:
>>>>     > > >
>>>>     > >
>>>>     http://lists.digium.com/pipermail/asterisk-video/2007-September/001257.html
>>>>     > > >
>>>>     > > > One solution is to reencode every video with a low bitrate
>>>>     and use a
>>>>     > > > fixed bitrate encoder (ffmpeg is dynamic bitrate for
>>>>     example). I had the
>>>>     > > > impression though that this still doesn't always work,
>>>>     especially when
>>>>     > > > the UMTS connection is not too good...
>>>>     > > >
>>>>     > > > My idea is that we check each time before sending a keyframe
>>>>     if there is
>>>>     > > > already a new keyframe in the queue. If this is the case, we
>>>>     jump to the
>>>>     > > > most recent keyframe and just drop the data before that.
>>>>     > > >
>>>>     > > > What do you think about this? How hard to implement?
>>>>     > > >
>>>>     > > > Regards,
>>>>     > > > Thomas F
>>>>     > > >
>>>>     > > >
>>>>     > > >
>>>>     > > > Am Montag, den 24.09.2007, 11:15 +0200 schrieb Thomas Z.:
>>>>     > > > > Hello,
>>>>     > > > >
>>>>     > > > > We have the problem, that a converted mp4 file is
>>>>     asynchronous via 3g
>>>>     > > > > network.
>>>>     > > > > If we play the file on a pc, everything is synchron.
>>>>     > > > > We tried already to reduce the quality and framerate of
>>>>     the video. But
>>>>     > > > > nothing helps.
>>>>     > > > >
>>>>     > > > > Is it a problem with mp4play or with the video?
>>>>     > > > >
>>>>     > > > > What can we do to get the audio and video synchronized via
>>>>     3g network?
>>>>     > > > >
>>>>     > > > > Thank you,
>>>>     > > > > best regards
>>>>     > > > > Thomas
>>>>     > > > >
>>>>     > > > >
>>>>     > > > >
>>>>     > > > > _______________________________________________
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>>>>     > > > --
>>>>     > > > www.ViiF.de - your Mobile Video Community
>>>>     > > >
>>>>     > > > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>>>>     > > >
>>>>     > > > Thomas Frieling - IT Development
>>>>     > > > ViiF Mobile Video GmbH, Poststr. 21-22, 10178 Berlin
>>>>     > > > Cell: +49 (0) 173 63 62 62 3
>>>>     > > >
>>>>     > > > mailto:thomas at ViiF.de
>>>>     > > >
>>>>     > > > Sitz: Berlin, Amtgericht Berlin-Charlottenburg, HRB: 108350B
>>>>     > > >
>>>>     > > > Geschäftsführer: Daniel Höpfner, Steffen Brünn
>>>>     > > >
>>>>     > > > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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>>>>     > >
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>>  
>>
>>
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> 
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