add latency to audio in gstrtpbin
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add latency to audio in gstrtpbin
 
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Hi,
If I have a pipeline using gstrtpbin (similar to the example in the documentation) to send audio and video, what is the best/most reliable way of adding latency to the audio? Would gst_event_new_latency work (and if so, how), or am I missing its intent: http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency Basically my concern is that if video capture is too slow, can I manually adjust the audio latency to match. Best, Tristan -- Tristan Matthews Société des arts technologiques [SAT] email: [hidden email] web: http://www.music.mcgill.ca/~tmatthews ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: add latency to audio in gstrtpbin
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On Tue, 2008-09-02 at 17:38 -0400, Tristan Matthews wrote:
> Hi, > > If I have a pipeline using gstrtpbin (similar to the example in the > documentation) to send audio and video, what is the best/most reliable > way of adding latency to the audio? Would gst_event_new_latency work > (and if so, how), or am I missing its intent: > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > Basically my concern is that if video capture is too slow, can I > manually adjust the audio latency to match. latency event is inform pipeline elements of the overlay latency of the pipeline. -- Olivier Crête [hidden email] Collabora Ltd ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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Re: add latency to audio in gstrtpbin
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In reply to this post by Tristan Matthews-2
Hi, Tristan:
You should not adjust video/audio latency manually because RTP provides a mechanism to accomplish this, called `lip-synchronization'. Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for more details. These will help you a lot. Eric Zhang 2008/9/3 Tristan Matthews <[hidden email]> Hi, ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: add latency to audio in gstrtpbin
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Thanks for the feedback Eric. I had seen the RFC before but that book
looks pretty useful as well. My concern isn't that the AV will get out of sync in gstreamer/gstrtpbin, but rather that it will be out of sync immediately at the capture stage, i.e. if i have video input from a camera with more latency than a separate audio source. An example scenario is given here: http://chris.pirillo.com/2007/07/11/audio-video-capture/ where you have video of someone clapping and you have to adjust the delay to make the sound and image of the clap line up. I would need to be able to manually adjust the latency to ensure the video and audio are sync'd. Should I just put the audio through a Ladspa-delay or is their a better solution in gstreamer? -T Eric Zhang wrote: > Hi, Tristan: > > You should not adjust video/audio latency manually because RTP > provides a mechanism to accomplish this, called `lip-synchronization'. > Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for > more details. These will help you a lot. > > Eric Zhang > > 2008/9/3 Tristan Matthews <[hidden email] <mailto:[hidden email]>> > > Hi, > > If I have a pipeline using gstrtpbin (similar to the example in the > documentation) to send audio and video, what is the best/most reliable > way of adding latency to the audio? Would gst_event_new_latency work > (and if so, how), or am I missing its intent: > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > Basically my concern is that if video capture is too slow, can I > manually adjust the audio latency to match. > > Best, > > Tristan > > -- > Tristan Matthews > Société des arts technologiques [SAT] > email: [hidden email] <mailto:[hidden email]> > web: http://www.music.mcgill.ca/~tmatthews > <http://www.music.mcgill.ca/%7Etmatthews> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > <http://moblin-contest.org/redirect.php?banner_id=100&url=/> > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > <mailto:[hidden email]> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Tristan Matthews Société des arts technologiques [SAT] email: [hidden email] web: http://www.music.mcgill.ca/~tmatthews ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: add latency to audio in gstrtpbin
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On Thu, 2008-09-04 at 11:34 -0400, Tristan Matthews wrote:
> Thanks for the feedback Eric. I had seen the RFC before but that book > looks pretty useful as well. > My concern isn't that the AV will get out of sync in > gstreamer/gstrtpbin, but rather that it will be out of sync immediately > at the capture stage, i.e. if i have video input from a camera with more > latency than a separate audio source. An example scenario is given here: > http://chris.pirillo.com/2007/07/11/audio-video-capture/ > where you have video of someone clapping and you have to adjust the > delay to make the sound and image of the clap line up. > I would need to be able to manually adjust the latency to ensure the > video and audio are sync'd. Should I just put the audio through a > Ladspa-delay or is their a better solution in gstreamer? > This situation can not occur in gstreamer when you have a well written source element that generates correct timestamps and reports its latency correctly. You could write an element that applies an offset to all timestamps on the buffers. Wim > -T > > Eric Zhang wrote: > > Hi, Tristan: > > > > You should not adjust video/audio latency manually because RTP > > provides a mechanism to accomplish this, called `lip-synchronization'. > > Refer to RFC 3550 or book `RTP: Video and Audio for the Internet' for > > more details. These will help you a lot. > > > > Eric Zhang > > > > 2008/9/3 Tristan Matthews <[hidden email] <mailto:[hidden email]>> > > > > Hi, > > > > If I have a pipeline using gstrtpbin (similar to the example in the > > documentation) to send audio and video, what is the best/most reliable > > way of adding latency to the audio? Would gst_event_new_latency work > > (and if so, how), or am I missing its intent: > > http://gstreamer.freedesktop.org/data/doc/gstreamer/stable/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-latency > > > > Basically my concern is that if video capture is too slow, can I > > manually adjust the audio latency to match. > > > > Best, > > > > Tristan > > > > -- > > Tristan Matthews > > Société des arts technologiques [SAT] > > email: [hidden email] <mailto:[hidden email]> > > web: http://www.music.mcgill.ca/~tmatthews > > <http://www.music.mcgill.ca/%7Etmatthews> > > > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's > > challenge > > Build the coolest Linux based applications with Moblin SDK & win > > great prizes > > Grand prize is a trip for two to an Open Source event anywhere in > > the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > <http://moblin-contest.org/redirect.php?banner_id=100&url=/> > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > <mailto:[hidden email]> > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------ > > > > ------------------------------------------------------------------------- > > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > > Build the coolest Linux based applications with Moblin SDK & win great prizes > > Grand prize is a trip for two to an Open Source event anywhere in the world > > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > [hidden email] > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: add latency to audio in gstrtpbin
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On Thu, Sep 4, 2008 at 1:54 PM, Wim Taymans <[hidden email]> wrote:
AFAIK, This element already exists and is called GstShift from gentrans plugins
-- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: add latency to audio in gstrtpbin
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In reply to this post by Wim Taymans
Hi, Tristan:
I agree with Wim. According to your requirement and problems, this is not the gstreamer or the network makes the audio/video out of sync. So you should apply different latency to your video and audio. Try to set `latency' property to your video and audio RTP stream or just write a plugin/probe to adjust the timestamp manually. Eric 2008/9/5 Wim Taymans <[hidden email]>
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