[vlc] two questions about transcoding in vlc

Marcelo Manzato manzato81 at yahoo.com.br
Mon Jan 30 18:58:10 CET 2006


Hi,

I'm implementing a video transcoding system using open
source tools such as vlc, mp4live and darwin streaming
server.

I'm using mp4live to capture video from a live video
camera, encode in a mpeg4 format and transmit via rtp
to darwin. Then, clients may access the content by
accessing via rtsp an sdp file previously generated. 

The transcoding process happens by the vlc application
accessing the sdp file in a same way - via rtsp
protocol. Then the transcoding process is done and the
modified content is transmitted back to darwin via
rtp, where a new sdp file for the modified content is
available to be accessed.

The darwin server, the vlc application and the mp4live
broadcaster are running in different machines, but all
in the same LAN.

One question is that when I increase the bitrate of
the transcoder (vlc), the clients receive the
transcoded content with some encoding errors, mainly
when there are too many movements and scene changes in
the image. By checking the log of vlc, I found many
messages like this:

access_output_udp debug: packet has been sent too late
(...)

The bandwidth seems to be good enough, because the
tests are being made in a 100 Mbits/s local and
isolated LAN. The computer running the vlc seems to be
good enough as well, it is a Intel P4 3Ghz with 1 GB
of RAM.

Another question is that I have been checking the
latency of the transmission in this system, and
surprisely, I found that, using the same live video
source, the transmission of the transcoded content is
spending less time to arrive to the clients then the
original content - without transcoding (about 0.5
seconds less). Considering that the transcoding
process takes time, it should be the opposite,
shouldn't it? 

Has the vlc application some module that controls the
time in real time situations, such as transcoding of
live video sources, by dropping packets or other
mechanisms in order to keep the time up?

The configuration of my video/audio content source is
the following:
frame size: 640x480
video bitrate: 1500 kbits/s
fps: 30
mpeg4 (ffmpeg)
44100 Hz of sampling
audio bitrate: 128 kbits/s
aac (faac).

Thanks in advance.





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