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<!-- @(#) $Header: /cs/research/mice/starship/src/local/CVS_repository/vic/html/research.html,v 1.1.1.1 1998/02/18 18:03:54 ucacsva Exp $ (LBL)-->
<html>
<head><title>vic Research Projects</title></head>
<body>
<h2>vic: Research Projects</h2>
Vic was developed as a research vehicle for exploring
packet video and multimedia conferencing as part of
<a href=http://www-nrg.ee.lbl.gov/mccanne>Steven McCanne's</a>
dissertation research at <a href=http://www.berkeley.edu>U.C. Berkeley</a>.
While the first generation design is in place,
vic continues as an active research project.
Research continues on the application and system architecture itself,
as well as on several projects that have evolved from the
vic development project:
<ul>
<li><a href=#thesis>Layered Video</a>
<li><a href=#vgw>Video Gateway</a>
<li><a href=#confbus>The Conference Bus</a>
<li><a href=#fasth261>Hardware-assisted Intra-H.261</a>
<li><a href=#loadadapt>CPU Load Adaptation</a>
<!-- <li><a href=#vc>Virtual Classroom</a> -->
</ul>
<a name=thesis><hr></a>
<h3>Layered Video</h3>
Internet video broadcasts are currently carried out by transmitting
a ``single-layer'' stream at a nominal bandwidth to all active multicast
receivers in the network.
By using a single-layer compression algorithm, a uniform quality of
video is transmitted to all users even those situated along high
bandwidth paths from the source.
Low-bandwidth portions of the multicast distribution are flooded
resulting in poor quality delivered to a possibly large fraction
of the users. Alternatively, the bandwidth can be scaled back
at the source to prevent such congestion, but then high bandwidth
users receive unnecessarily low quality.
<p>
Our solution is to utilize a
layered source-coder in tandem with a layered transmission system.
The layered source-coder produces an embedded bit stream that
can be decomposed into arbitrary number of hierarchical flows.
These flows are distributed across multiple multicast groups,
allowing receivers to ``tune in'' to some subset of the flows
<a href=#deering>[1]</a>.
Each layer provides a progressively higher quality signal.
<p>
We utilize only mechanisms that presently exist in the Internet and
leverage off a mechanism called ``pruning'' in IP Multicast.
IP Multicast works by constructing a simplex distribution tree
from each source subnet. A source transmits a packet with
an ``IP group address'' destination
and the multicast routers forward a copy of this packet along
each link in the distribution tree. When a destination subnet
has no active receivers for a certain group address, the last-hop
router sends a ``prune'' message back up the distribution tree,
which prevents intermediate routers from forwarding the unnecessary
traffic. Thus, each user can locally adapt to network capacity
by adjusting the number of multicast groups --- i.e., the number
of compression layers --- that they receive.
<p>
Our source-coder is a simple, low-complexity, wavelet-based algorithm
that can be run in software on standard workstations and PC's
(see <a href=#wavelet-book>[7]</a> for a comprehensive treatment
of wavelets applied to subband coding).
Conceptually, the algorithm works by conditionally replenishing
wavelet transform coefficients of each frame. By using a
wavelet decomposition with fairly short basis functions,
we can optimize the coder by carrying out conditional replenishment
in the pixel domain and then transforming and coding only
those blocks that need to be updated. The wavelet coefficients
are bit-plane coded using a representation similar to the
well-known zero-tree decomposition <a href=#shapiro>[2]</a>.
All zero-tree sets are computed in parallel with a single
bottom up traversal of the coefficient quad-tree, and all
layers of the bit-stream are computed in parallel using
a table-driven approach.
<p>
<a name=vgw><hr></a>
<h3>Video Gateway</h3>
While the layered compression algorithm and transmission system
described above provide an elegant and scalable solution to
the problem of heterogeneous video distribution, the technology is
new and will take some time to be deployed. In the meantime,
an alternative solution is the deployment of transcoding video gateways.
<p>
Using vic's software architecture and codec implementations
as a foundation, we have designed a new tool for video bandwidth
adaptation using efficient transcoding between video formats
<a href=#amir-mm95>[3]</a>. By placing an application-level gateway
near the entrance
to a low bandwidth environment, high-rate video can be
adapted for the lower bandwidth network through transcoding.
<p>
We have implemented our gateway design in an application
called <i>vgw</i> and used it to transcode high-quality JPEG video
from a
<a href=http://roger-rabbit.cs.berkeley.edu/classes/f95-CS298-5/home.html>
UCB Seminar</a> to low-rate H.261 video for the MBone.
The JPEG to H.261 conversion process uses
a highly optimized algorithm that avoids DCT computations by
manipulating data entirely within the transform domain.
<p>
<a href=http://http.cs.berkeley.edu:80/~elan/>Elan Amir</a> has continued
work on the gateway architecture as part of his Masters thesis, evolving
our prototype into a fully-functional application and integrating it
into other environments such as
<a href=http://infopad.EECS.Berkeley.EDU/>InfoPad</a> and
<a href=http://HTTP.CS.Berkeley.EDU/~fox/glomop/>GloMop</a>.
<p>
<a name=confbus><hr></a>
<h3>The Conference Bus</h3>
In July 1992, we added a multicast messaging protocol
to vat for interprocess communication in order to orchestrate ownership
of exclusive-access audio devices. The protocol was completely
decentralized and stateless, allowing instances of vat to join
and leave the communication group without an involved
synchronization procedure. Since then, we have used this
extensible ``Conference Bus'' abstraction to implement voice switched
windows, where vic uses cues from vat to switch a viewing
window to whomever is talking.
<p>
We're currently working on a floor control tool, where the media
applications are coordinated over the Conference Bus via remote
commands from the floor control tool.
All of the LBL MBone tools have the ability to ``mute'' or ignore
a network media source, and the disposition of this mute control
can be controlled via the Conference Bus, which the floor
control tool can use to effect a moderation policy.
One possible model is that each
participant in the session follows the direction of a
well-known (session-defined) moderator. The moderator can
give the floor to a participant by multicasting a <em> takes-floor</em>
directive with that participant's RTP CNAME.
Locally, each receiver then mutes all participants except the
one that holds the floor.
<p>
Cross-media synchronization can also be
carried out over the Conference Bus. Each real-time
application induces a buffering delay, called the <em> playback</em>
point, to adapt to packet delay variations <a href=#jacobson>[4]</a>.
This playback point can be adjusted to synchronize across media.
By broadcasting ``synchronize'' messages across
the Conference Bus, the different media can compute the maximum
of all advertised playout delays. This maximum is then
used in the delay-adaptation algorithm.
<p>
We've also used the
Conference Bus to implement remote manipulation of
graphical overlays in a vic video stream. This allows
a ``title-generator'' to be easily prototyped as separate
tool from vic.
<p>
<a href=http://http.cs.berkeley.edu:80/~elan/>Elan Amir</a>
has extended vgw with a Conference Bus interface, using
a split implementation where the transcoding engine
is a bare application with a tcl/tk user-interface
that configures the engine over the Conference Bus.
By using the Conference Bus, vgw automatically inherits the
flexibility of our existing conference coordination framework.
For example, vgw can be configured to dedicate extra bandwidth
to a video stream with an active speaker in the same way
vic can be configured to switch windows to the current speaker.
<p>
<a name=fasth261><hr></a>
<h3>Hardware-assisted Intra-H.261</h3>
<a href=http://www-nrg.ee.lbl.gov/kfall/>Kevin Fall</a>
has integrated the efficient JPEG-to-H.261 algorithm from vgw
into vic's capture model, allowing existing hardware JPEG codecs
to serve as H.261 sources. Rather than encode H.261 from
scratch, a JPEG codec can be used as a <i>DCT Engine</i>
by entropy decoding the compressed bitstream without performing
reverse DCT's <a href=#mccanne-ietf>[5]</a>.
The intermediate transform coefficients can then be encoded directly
as H.261, avoiding DCT computations entirely. Since Huffman
encoding and decoding is much cheaper than the DCT computations,
the algorithm runs fast.
<p>
<a name=loadadapt><hr></a>
<h3>CPU Load Adaptation</h3>
Vic currently processes packets as soon as they arrive and renders
frames as soon as a complete frame is received. Under this scheme,
when a receiver can't keep up with a high-rate source, quality
degrades drastically because packets get dropped indiscriminately
at the kernel input buffer. A better approach is to gracefully
adapt to offered load by adjusting a compute-scalable
decoder <a href=#fall>[6]</a>
<p>
A large component of the decoding CPU budget goes into rendering
frames, whether copying bits to the frame buffer, performing color
space conversion, or carrying out a dither. Accordingly, the
vic rendering modules were designed so that their load can be
adapted by running the rendering process at a rate lower than
the incoming frame rate. Thus, a 10 f/s H.261 stream can be
rendered, say, at 4 f/s if necessary. Alternatively, the decoding
process itself can be scaled (if possible). For example, our
prototype layered coder can run faster by processing
a reduced number layers at the cost of lower image quality.
<p>
While the hooks for scaling the decoder process are in place,
the control algorithm isn't. We are currently working on
approaches to measuring load and the control algorithm for
updating the load-scaling parameters.
<p>
<!--
<a name=vc><hr></a>
<h3>The Virtual Classroom</h3>
-->
<hr>
<h3>References</h3>
<p>
<dl compact>
<dt><a name=deering><strong>[1]</strong></a>
<dd>
Deering, S., ``Re: hierarchical encoding?''.
Email message sent to the
<a href=ftp:://cs.ucl.ac.uk/darpa/IDMR/idmr-archive.Z>
Inter-Domain Multicast Routing (IDMR) mailing list</a>, Jan 1994.
<dt><a name=shapiro><strong>[2]</strong></a>
<dd>
Shapiro, J. M.,
``Embedded Image Coding Using Zerotrees of Wavelet Coefficients,''
IEEE Transactions on Signal Processing,
Vol. 41, No. 12, Dec. 1993, pp. 3445-3462.
<dt><a name=amir-mm95><strong>[3]</strong></a>
<dd>
Amir, E., McCanne, S., and Zhang, H.,
<a href="http://www.cs.berkeley.edu/~elan/articles/pub/vgw.ps">
An Application-level Video Gateway.</a>
To appear in ACM Multimedia '95.
<dt><a name=jacobson><strong>[4]</strong></a>
<dd>
Jacobson, V.
<a href=ftp://cs.ucl.ac.uk/darpa/vjtut.ps.Z>
SIGCOMM '94 Tutorial: Multimedia conferencing on the Internet</a>, Aug. 1994.
<dt><a name=mccanne-ietf><strong>[5]</strong></a>
<dd>
McCanne, S.,
<a href="http://www-nrg.ee.lbl.gov/mccanne/talks/avt-ietf-dec94.ps.gz">
vic and RTPv2</a>,
Audio-Video Transport Working Group, IETF 31.
December 6, 1994. San Jose, CA.
<dt><a name=fall><strong>[6]</strong></a>
<dd>
Fall, K., Pasquale, J., and McCanne, S.
<strong>
Workstation Video Playback Performance with Competitive Process Load.</strong>
In Proceedings of the Fifth International Workshop
on Network and OS Support for Digital Audio and Video.
April, 1995. Durham, NH.
<dt><a name=wavelet-book><strong>[7]</strong></a>
<dd>
Veterrli, M. and Kovacevic, J.
<strong>
<a href=http://gabor.eecs.berkeley.edu/~martin/book.html>
Wavelets and Subband Coding</a></strong>
Prentice-Hall, 1995. ISBN: 0-13-097080-8.
</dl>
<a href=vic.html>[Return to main vic page]</a>
</body>
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