rtp: update README, fix some typos, mention gstrtpbin
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@ -165,7 +165,7 @@ between the RTP and GST timestamps. This information is used by a session
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manager to create SR reports. The NTP time in the report will contain the
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manager to create SR reports. The NTP time in the report will contain the
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running time converted to NTP time and the corresponding RTP timestamp.
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running time converted to NTP time and the corresponding RTP timestamp.
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Not that at the sender side, the RTP and GStreamer timestamp both increment at
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Note that at the sender side, the RTP and GStreamer timestamp both increment at
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the same rate, the sender rate. This rate depends on the global pipeline clock
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the same rate, the sender rate. This rate depends on the global pipeline clock
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of the sender.
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of the sender.
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@ -225,7 +225,7 @@ and will apply the drift correction to the GStreamer timestamp before pushing
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the buffer downstream. The result is that the depayloader receives a smoothed
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the buffer downstream. The result is that the depayloader receives a smoothed
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GStreamer timestamp on the RTP packet, which is copied to the depayloaded data.
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GStreamer timestamp on the RTP packet, which is copied to the depayloaded data.
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The following pipeline illustrates the sender with a jitterbuffer.
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The following pipeline illustrates a receiver with a jitterbuffer.
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gst-launch udpsrc caps="application/x-rtp, media=(string)video,
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gst-launch udpsrc caps="application/x-rtp, media=(string)video,
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clock-rate=(int)90000, encoding-name=(string)H263-1998" !
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clock-rate=(int)90000, encoding-name=(string)H263-1998" !
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@ -326,6 +326,17 @@ Some gst-launch lines:
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The caps on the udpsinks can be retrieved when the server pipeline prerolled to
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The caps on the udpsinks can be retrieved when the server pipeline prerolled to
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PAUSED.
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PAUSED.
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The above pipeline sets sync=false on the audio and video sink which means that
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no synchronisation will be performed in the sinks, they play the data when it
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arrives. If you want to enable synchronisation in the sinks it is highly
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recommended to use a gstrtpjitterbuffer after the udpsrc elements.
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Even when sync is enabled, the two different streams will not play synchronised
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against eachother because the receiver does not have enough information to
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perform this task. For this you need to add the gstrtpbin element in both the
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sender and receiver pipeline and use additional sources and sinks to transmit
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RTCP packets used for inter-stream synchronisation.
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The caps on the receiver side can be set on the UDP source elements when the
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The caps on the receiver side can be set on the UDP source elements when the
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pipeline went to PAUSED. In that state no data is received from the UDP sources
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pipeline went to PAUSED. In that state no data is received from the UDP sources
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as they are live sources and only produce data in PLAYING.
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as they are live sources and only produce data in PLAYING.
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