U.S. patent application number 11/928741 was filed with the patent office on 2008-05-01 for data transmission.
This patent application is currently assigned to NOKIA CORPORATION. Invention is credited to Miska HANNUKSELA.
Application Number | 20080104647 11/928741 |
Document ID | / |
Family ID | 8554549 |
Filed Date | 2008-05-01 |
United States Patent
Application |
20080104647 |
Kind Code |
A1 |
HANNUKSELA; Miska |
May 1, 2008 |
DATA TRANSMISSION
Abstract
A data transmission system (10) is provided for transmitting a
data signal between a server (12, 14, 16, 18) and a communications
terminal (20, 22) over a transmission link having a variable
bandwidth. The data signal is a scalable compressed multimedia clip
comprising a sequence of images. Each image has a base layer and a
number of enhancement layers. To cope with variability in the
bandwidth, the sequence is re-ordered so that the base layers have
greater safety times than the enhancement layers. This allows all
of the base layers to arrive in time to be played at the expense of
the enhancement layers.
Inventors: |
HANNUKSELA; Miska; (Tampere,
FI) |
Correspondence
Address: |
PERMAN & GREEN
425 POST ROAD
FAIRFIELD
CT
06824
US
|
Assignee: |
NOKIA CORPORATION
Keilalahdentie 4
Espoo
FI
02150
|
Family ID: |
8554549 |
Appl. No.: |
11/928741 |
Filed: |
October 30, 2007 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
|
09560217 |
Apr 28, 2000 |
7289506 |
|
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11928741 |
Oct 30, 2007 |
|
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Current U.S.
Class: |
725/114 ;
375/E7.011; 375/E7.09; 375/E7.091 |
Current CPC
Class: |
H04N 19/37 20141101;
H04N 19/31 20141101; H04N 21/234381 20130101; H04N 21/234327
20130101; H04N 21/2662 20130101; H04N 21/234354 20130101; H04N
21/6131 20130101; H04N 21/2402 20130101; H04N 21/234363 20130101;
H04N 21/6377 20130101 |
Class at
Publication: |
725/114 |
International
Class: |
H04N 7/173 20060101
H04N007/173 |
Foreign Application Data
Date |
Code |
Application Number |
Apr 29, 1999 |
FI |
990970 |
Claims
1-26. (canceled)
27. A server for streaming a data signal comprising a sequence of
data units, the server comprising: a re-ordering unit for changing
an order of primary and secondary data units in the sequence of
data units by exchanging a secondary data unit that precedes a
primary data unit in an original data unit decoding order with the
primary data unit.
28. A server according to claim 1, wherein the re-ordering unit is
configured to re-order a primary and a secondary data unit
representing respectively a base layer and at least one enhancement
layer of the data signal.
29. A server according to claim 1, wherein the re-ordering unit is
configured to re-order data units based at least in part on a
prioritization of the data units.
30. A server according to claim 1, wherein the re-ordering unit is
configured to re-order a primary and a secondary data unit
representing respectively a base layer of a first frame of a video
sequence and an enhancement layer of a second frame of the video
sequence, the second frame preceding the first frame in the
original decoding order.
31. A server according to claim 1, wherein the re-ordering unit is
configured to re-order data units of a data signal representing a
video sequence.
32. A method for streaming a data signal comprising a sequence of
data units, the method comprising: changing an order of primary and
secondary data units in the sequence of data units by exchanging a
secondary data unit that precedes a primary data unit in an
original data unit decoding order with the primary data unit.
33. A method according to claim 32, wherein changing the order of
primary and secondary data units comprises: exchanging a primary
and a secondary data unit representing respectively a base layer
and at least one enhancement layer of the data signal.
34. A method according to claim 32, wherein changing the order of
primary and secondary data units comprises: re-ordering data units
based at least in part on a prioritization of the data units.
35. A method according to claim 32, wherein changing the order of
primary and secondary data units comprises: re-ordering a primary
and a secondary data unit representing respectively a base layer of
a first frame of a video sequence and an enhancement layer of a
second frame of the video sequence, the second frame preceding the
first frame in the original decoding order.
36. A method according to claim 32, wherein the data units of a
data signal are representative of a video sequence.
37. A client device, the device comprising: an input for receiving
a sequence of data units; a re-ordering unit for changing an order
of primary and secondary data units in the received sequence of
data units by exchanging a primary data unit that precedes a
secondary data unit in the received sequence of data units with the
secondary data unit to recover a decoding order of the data units;
and a decoder for decoding the reordered data units.
38. A client according to claim 37, wherein the re-ordering unit is
configured to re-order a primary and a secondary data unit
representing respectively a base layer and at least one enhancement
layer of the data signal.
39. A client according to claim 37, wherein the re-ordering unit is
configured to re-order a primary and a secondary data unit
representing respectively a base layer of a first frame of a video
sequence and an enhancement layer of a second frame of the video
sequence, the second frame preceding the first frame in the
original decoding order.
40. A client according to claim 37, wherein the re-ordering unit is
configured to re-order data units of a data signal representing a
video sequence.
41. A method comprising: receiving a sequence of data units;
changing an order of primary and secondary data units in the
received sequence of data units by exchanging a primary data unit
that precedes a secondary data unit in the received sequence of
data units with the secondary data unit to recover a decoding order
of the data units; and decoding the reordered data units.
42. A method according to claim 41, wherein changing the order of
primary and secondary data units comprises: re-ordering a primary
and a secondary data unit representing respectively a base layer
and at least one enhancement layer of the data signal.
43. A method according to claim 41, wherein changing the order of
primary and secondary data units comprises: re-ordering a primary
and a secondary data unit representing respectively a base layer of
a first frame of a video sequence and an enhancement layer of a
second frame of the video sequence, the second frame preceding the
first frame in the original decoding order.
44. A method according to claim 41, wherein the data units of a
data signal are representative of a video sequence.
45. A computer-readable medium embodying a computer program
executable by one or more processors, comprising: computer code for
receiving a sequence of data units; computer code for changing an
order of primary and secondary data units in the received sequence
of data units by exchanging a primary data unit that precedes a
secondary data unit in the received sequence of data units with the
secondary data unit to recover a decoding order of the data units;
and computer code for decoding the reordered data units.
46. A computer-readable medium according to claim 45, wherein
computer code changing the order of primary and secondary data
units comprises: computer code for re-ordering a primary and a
secondary data unit representing respectively a base layer and at
least one enhancement layer of the data signal.
47. A computer-readable medium according to claim 45, wherein
computer code for changing the order of primary and secondary data
units comprises: computer code for re-ordering a primary and a
secondary data unit representing respectively a base layer of a
first frame of a video sequence and an enhancement layer of a
second frame of the video sequence, the second frame preceding the
first frame in the original decoding order.
48. A computer-readable medium according to claim 45, wherein the
data units of a data signal are representative of a video sequence.
Description
FIELD OF THE INVENTION
[0001] This invention relates to data transmission and is
particularly, but not exclusively, related to transmission of data
representative of picture sequences, such as video. It is
particularly suited to transmission over links susceptible to
errors and loss of data.
BACKGROUND OF THE INVENTION
[0002] During the past few years, the amount of multimedia content
available through the Internet has increased considerably. Since
data delivery rates to mobile terminals are becoming high enough to
enable such terminals to be able to retrieve multimedia content, it
is becoming desirable to enable mobile terminals to retrieve video
and other multimedia content from the Internet. An example of a
high-speed data delivery system is the upcoming GSM phase 2+.
[0003] The term multimedia as used herein applies to both sound and
pictures, to sound only and to pictures only. Sound may include
speech and music.
[0004] Network traffic through the Internet is based on a transport
protocol called the Internet Protocol (IP). IP is concerned with
transporting data packets from one location to another. It
facilitates the routing of packets through intermediate gateways,
that is, it allows data to be sent to machines that are not
directly connected in the same physical network. The unit of data
transported by the IP layer is called an IP datagram. The delivery
service offered by IP is connectionless, that is IP datagrams are
routed around the Internet independently of each other. Since no
resources are permanently committed within the gateways to any
particular connection, the gateways may occasionally have to
discard datagrams because of lack of buffer space or other
resources. Thus, the delivery service offered by IP is a best
effort service rather than a guaranteed service.
[0005] Internet multimedia is typically streamed over the User
Datagram Protocol (UDP), the Transmission Control Protocol (TCP) or
the Hypertext Transfer Protocol (HTTP).
[0006] UDP is a connectionless lightweight transport protocol. It
offers very little above the service offered by IP. Its most
important function is to deliver datagrams between specific
transport endpoints. Consequently, the transmitting application has
to take care of how to packetize data to datagrams. Headers used in
UDP contain a checksum that allows the UDP layer at the receiving
end to check the validity of the data. Otherwise, degradation of IP
datagrams will in turn affect UDP datagrams. UDP does not check
that the datagrams have been received, does not retransmit missing
datagrams, nor does it guarantee that the datagrams are received in
the same order as they were transmitted.
[0007] UDP introduces a relatively stable throughput having a small
delay since there are no retransmissions. Therefore it is used in
retrieval applications to deal with the effect of network
congestion and to reduce delay (and jitter) at the receiving end.
However, the client must be able to recover from packet losses and
possibly conceal lost content. Even with reconstruction and
concealment, the quality of a reconstructed clip suffers somewhat.
On the other hand, playback of the clip is likely to happen in
real-time without annoying pauses. Firewalls, whether in a company
or elsewhere, may forbid the usage of UDP because it is
connectionless.
[0008] TCP is a connection-orientated transport protocol and the
application using it can transmit or receive a series of bytes with
no apparent boundaries as in UDP. The TCP layer divides the byte
stream into packets, sends the packets over an IP network and
ensures that the packets are error-free and received in their
correct order. The basic idea of how TCP works is as follows. Each
time TCP sends a packet of data, it starts a timer. When the
receiving end gets the packet, it immediately sends an
acknowledgement back to the sender. When the sender receives the
acknowledgement, it knows all is well, and cancels the timer.
[0009] However, if the IP layer loses the outgoing segment or the
return acknowledgement, the timer at the sending end will expire.
At this point, the sender will retransmit the segment. Now, if the
sender waited for an acknowledgement for each packet before sending
the next one, the overall transmission time would be relatively
long and dependent on the round-trip delay between the sender and
the receiver. To overcome this problem, TCP uses a sliding window
protocol that allows several unacknowledged packets to be present
in the network. In this protocol, an acknowledgement packet
contains a field filled with the number of bytes the client is
willing to accept (beyond the ones that are currently
acknowledged). This window size field indicates the amount of
buffer space available at the client for storage of incoming data.
The sender may transmit data within the limit indicated by the
latest received window size field. The sliding window protocol
means that TCP effectively has a slow start mechanism. At the
beginning of a connection, the very first packet has to be
acknowledged before the sender can send the next one. Typically,
the client then increases the window size exponentially. However,
if there is congestion in the network, the window size is decreased
(in order to avoid congestion and to avoid receive buffer
overflow). The details how the window size is changed depend on the
particular TCP implementation in use.
[0010] A multimedia content creation and retrieval system is shown
in FIG. 1. The system has one or more media sources, for example a
camera and a microphone. Alternatively, multimedia content can also
be synthetically created without a natural media source, for
example animated computer graphics and digitally generated music.
In order to compose a multimedia clip consisting of different media
types, such as video, audio, text, images, graphics and animation,
raw data captured from the sources are edited by an editor.
Typically the storage space taken up by raw (uncompressed)
multimedia data is huge. It can be megabytes for a video sequence
which can include a mixture of different media, for example
animation. In order to provide an attractive multimedia retrieval
service over low bit rate channels, for example 28.8 kbps and 56
kbps, multimedia clips are compressed in the editing phase. This
typically occurs off-line. The clips are then handed to a
multimedia server. Typically, a number of clients can access the
server over one or more networks. The server is able to respond to
the requests presented by the clients. The main task of the server
is to transmit a desired multimedia clip to the client which the
client decompresses and plays. During playback, the client utilizes
one or more output devices, such as a screen and a loudspeaker. In
some circumstances, clients are able to start playback while data
are still being downloaded.
[0011] It is convenient to deliver a clip by using a single channel
which provides a similar quality of service for the entire clip.
Alternatively different channels can be used to deliver different
parts of a clip, for example sound on one channel and pictures on
another. Different channels may provide different qualities of
service. In this context, quality of service includes bit rate,
loss or bit error rate and transmission delay variation.
[0012] In order to ensure multimedia content of a sufficient
quality is delivered, it is provided over a reliable network
connection, such as TCP, which ensures that received data are
error-free and in the correct order. Lost or corrupted protocol
data units are retransmitted. Consequently, the channel throughput
can vary significantly. This can even cause pauses in the playback
of a multimedia stream whilst lost or corrupted data are
retransmitted. Pauses in multimedia playback are annoying.
[0013] Sometimes retransmission of lost data is not handled by the
transport protocol but rather by some higher-level protocol. Such a
protocol can select the most vital lost parts of a multimedia
stream and request the retransmission of those. The most vital
parts can be used for prediction of other parts of the stream, for
example.
[0014] In order to understand the invention better, descriptions of
the elements of the retrieval system, namely the editor, the server
and the client, are set out below.
[0015] A typical sequence of operations carried out by the
multimedia clip editor is shown in FIG. 2. Raw data are captured
from one or more data sources. Capturing is done using hardware,
device drivers dedicated to the hardware and a capturing
application which controls the device drivers to use the hardware.
Capturing hardware may consist of a video camera connected to a PC
video grabber card, for example. The output of the capturing phase
is usually either uncompressed data or slightly compressed data
with irrelevant quality degradations when compared to uncompressed
data. For example, the output of a video grabber card could be in
an uncompressed YUV 4:2:0 format or in a motion-JPEG format. The
YUV colour model and the possible sub-sampling schemes are defined
in Recommendation ITU-R BT.601-5 "Studio Encoding Parameters of
Digital Television for Standard 4:3 and Wide-Screen 16:9 Aspect
Ratios". Relevant digital picture formats such as CIF, QCIF and
SQCIF are defined in Recommendation ITU-T H.261 "Video Codec for
Audiovisual Services at p.times.64 kbits" (section 3.1 "Source
Formats").
[0016] During editing separate media tracks are tied together in a
single timeline. It is also possible to edit the media tracks in
various ways, for example to reduce the video frame rate. Each
media track may be compressed. For example, the uncompressed YUV
4:2:0 video track could be compressed using ITU-T recommendation
H.263 for low bit rate video coding. If the compressed media tracks
are multiplexed, they are interleaved so that they form a single
bitstream. This clip is then handed to the multimedia server.
Multiplexing is not essential to provide a bitstream. For example,
different media components such as sounds and images may be
identified with packet header information in the transport layer.
Different UDP port numbers can be used for different media
components.
[0017] A typical sequence of operations carried out by the
multimedia server is shown in FIG. 3. Typically multimedia servers
have two modes of operation; they deliver either pre-stored
multimedia clips or a live (real-time) multimedia stream. In the
first mode, clips are stored in a server database, which is then
accessed on-demand by the server. In the second mode, multimedia
clips are handed to the server as a continuous media stream that is
immediately transmitted to clients. Clients control the operation
of the server by an appropriate control protocol being at least
able to select a desired media clip. In addition, servers may
support more advanced controls. For example, clients may be able to
stop the transmission of a clip, to pause and resume transmission
of a clip, and to control the media flow in case of a varying
throughput of the transmission channel in which case the server
must dynamically adjust the bitstream to fit into the available
bandwidth.
[0018] A typical sequence of operations carried out by the
multimedia retrieval client is shown in FIG. 4. The client gets a
compressed and multiplexed media clip from a multimedia server. The
client demultiplexes the clip in order to obtain separate media
tracks. These media tracks are then decompressed to provide
reconstructed media tracks which are played out with output
devices. In addition to these operations, a controller unit is
provided to interface with end-users, that is to control playback
according to end-user input and to handle client-server control
traffic. It should be noted that the
demultiplexing-decompression-playback chain can be done on a first
part of the clip while still downloading a subsequent part of the
clip. This is commonly referred to as streaming. An alternative to
streaming is to download the whole clip to the client and then
demultiplex it, decompress it and play it.
[0019] A typical approach to the problem of varying throughput of a
channel is to buffer media data in the client before starting the
playback and/or to adjust the transmitted bit rate in real-time
according to channel throughput statistics.
[0020] One way of tackling the problem of pauses is by using
dynamic bit rate adjustment in the multimedia server. However, the
server may not react to network congestion sufficiently quickly to
avoid pausing in the client. In addition, the server cannot control
the retransmission mechanism of TCP (or other underlying protocols,
such as IP).
[0021] Even if dynamic bit rate adjustment is used, the client has
to do some initial buffering in any case to avoid delivery delays
caused by re-transmission. If a constant channel bit rate is
assumed, one can calculate the point in time at which a data unit
is supposed to have been completely received. In addition, one can
calculate the point in time by which a data unit is supposed to
have been played. The time difference between these two points in
time is referred to as the safety time. Another way of defining the
safely time is to state that it is the maximum time between two
consecutively received data units which does not cause pausing in
the playback.
[0022] When calculating the safety times for a clip, each data unit
has to be considered separately. The calculations assume that no
throughput drops occur before the data unit that is currently being
processed. If the maximum throughput of the channel is equal to the
average bit rate of the multimedia clip, the client cannot recover
from a drop in the amount of received bits after throughput has
dropped. The only way to guarantee some protection against
throughput drops is to buffer some data before starting playback.
If the channel stops providing data, the client can still play the
stream while there are data in a buffer. Thus, the average safety
time is approximately equal to the initial buffering time. Since
the bit rate of the clip may vary, the safety time also may vary
and the minimum safety time is thus equal to or less than the
initial buffering delay.
[0023] HTTP, the Hypertext Transfer Protocol, is the basis of the
World Wide Web (WWW). It is a simple protocol. A client establishes
a TCP connection to a server, issues a request, and reads back the
server's response. The server denotes the end of its response by
closing the connection. The arrangement of protocol layers is
typically HTTP on TCP which is on IP.
[0024] The most common HTTP request is called GET. The GET request
is associated with a universal resource identifier (URI) which
uniquely specifies the requested item. The server responds to the
GET request by returning the file corresponding to the specified
URI. The file returned by the server normally contains pointers
(hypertext links) to other files that can reside on other servers.
A user can therefore easily follow the links from file to file.
[0025] Servers used for Internet multimedia retrieval are either
dedicated multimedia servers or normal WWW servers.
[0026] Dedicated multimedia servers are typically capable of
transmission over HTTP, TCP and UDP protocols. They may be able to
readjust the contents of media clips dynamically to meet the
available network bandwidth and to avoid network congestion. They
may also support fast forward and fast rewind operations as well as
live multimedia streaming. They can provide a number of streams
simultaneously.
[0027] Multimedia servers based on normal WWW servers are also
referred to as server-less or HTTP multimedia solutions. The
multimedia clips are streamed over HTTP. Since this type of server
has no control over the contents of the stream, no flow (bandwidth)
control can be applied, and it cannot respond to network
congestion. Therefore, sudden pauses in the playback can occur.
Consequently there must be a relatively long initial buffering
delay in the client before starting the playback to avoid such
sudden pauses. Fast-forwarding a multimedia stream from a standard
WWW server is not possible. Live multimedia streaming must be
implemented using special tricks, such as Java programming.
[0028] When a streamed multimedia clip is received, a suitable
independent media player application or a browser plug-in can be
used to play it. Such multimedia players differ largely from
browser to browser. Newer browsers may have some integrated
plug-ins for the most popular streaming video players.
[0029] There are a number of different data transmission methods
available for transmitting data between mobile terminals and their
networks. One of the best known methods is GSM (Global System for
Mobile communications).
[0030] The current GSM data service called Circuit Switched Data
(CSD) offers a 9.6 kbps circuit-switched channel. It is intended
for GSM to provide a 14.4 kbps data channel having forward error
correction (FEC) and status information. High Speed Circuit
Switched Data (HSCSD) provides multiple 9.6 kbps or 14.4 kbps time
slots for a single user at the same time. There are symmetric and
asymmetric connections. In a symmetric connection, the air
interface resources are allocated symmetrically, and provide the
same data transmission rate in both directions. In an asymmetric
connection, different data rates are supported for up-link and for
down-link. However, asymmetric air interface connections are
applicable only in non-transparent mode (see below).
[0031] Circuit-switched GSM data systems, CSD and HSCSD, offer two
basic connection types, namely transparent (T) and non-transparent
(NT). These are distinguished by the way they correct transmission
errors. In a transparent connection, error correction is done
solely by a forward error correction mechanism provided by the
radio interface transmission scheme. The connection is considered
as a synchronous circuit. The available throughput is constant, and
the transmission delay is fixed. The transmitted data are likely to
contain bit inversion errors. In a non-transparent connection, the
GSM circuit connection is regarded as a packet (or frame) data flow
although the end-to-end service is circuit-switched. Each frame
includes redundancy bits to enable a receiver to detect remaining
errors. There are two sources of error, packet drops and corruption
due to interference in the radio frequency path. The latter can be
recovered by redundancy checking. The Radio Link Protocol (RLP) is
used to provide retransmission in case of remaining errors in
frame. If a frame is found to be correct, the receiver acknowledges
this fact. If it is found not to be correct, a negative
acknowledgement is sent and the indicated frame is retransmitted.
Consequently, a non-transparent connection is error-free but
throughput and transmission delay vary.
[0032] Other network types exist, for example GPRS (General Packet
Radio System). In GPRS the transmissions are truly packet
based.
[0033] A video sequence consists of a series of still pictures.
Video compression methods are based on reducing redundant and
perceptually irrelevant parts of video sequences. The redundancy in
video sequences can be categorised into spatial, temporal and
spectral redundancy. Spatial redundancy means the correlation
between neighbouring pixels. Temporal redundancy means that the
same objects appear in consecutive pictures. Reducing the temporal
redundancy reduces the amount of data required to represent a
particular image sequence and thus compresses the data. This can be
achieved by generating motion compensation data, which describe the
motion between the current and a previous (reference) picture. In
effect, the current picture is predicted from the previous one.
Spectral redundancy means the correlation between the different
colour components of the same picture.
[0034] Simply reducing the redundancy of the sequence does not
usually compress it enough. Therefore, some video encoders try to
reduce the quality of those parts of a video sequence which are
subjectively the least important. In addition, the redundancy of
the encoded bitstream is reduced by means of efficient lossless
coding of compression parameters and coefficients. The main
technique is to use variable length codes.
[0035] Video compression methods typically differentiate between
pictures that can use temporal redundancy reduction and those that
cannot. Compressed pictures, which do not use temporal redundancy
reduction methods, are usually called INTRA or I-frames whereas
temporally predicted pictures are called INTER or P-frames. In the
INTER frame case, the predicted (motion-compensated) picture is
rarely precise enough, and therefore a spatially compressed
prediction error picture is also associated with each INTER
frame.
[0036] Temporal scalability provides a mechanism for enhancing
perceptual quality by increasing the picture display rate. This is
achieved by taking a pair of consecutive reference pictures and
bi-directionally predicting a B-picture from either one or both of
them. The B-picture can then be played in sequence between the two
anchor pictures. This is illustrated in FIG. 5. Bi-directional
temporal prediction yields a more accurately predicted picture than
uni-directional prediction. Thus, for the same quantization level,
B-pictures yield increased compression as compared to forwardly
predicted P-pictures. B-pictures are not used as reference
pictures, that is other pictures are never predicted from them.
Since they can be discarded without impacting the picture quality
of future pictures, they provide temporal scalability. It should be
noted that while B-pictures provide better compression performance
than P-pictures, they are more complex to construct and require
more memory. Furthermore they introduce additional delays because
bi-directional interpolation requires both reference pictures to
have been received and additional calculations are required. In
addition, B-pictures require more side information in the
bitstream.
[0037] The term scalability refers to the capability of a
compressed sequence to be decoded at different data rates. In other
words, a scalable multimedia clip can be edited relatively easily
while it is compressed so that it can be streamed over channels
with different bandwidths and still be decoded and played back in
real-time.
[0038] Scalable multimedia is typically ordered so that there are
hierarchical layers of data. A base layer contains a basic
representation of the multimedia clip whereas enhancement layers
contain refinement data on top of underlying layers. Consequently,
the enhancement layers improve the quality of the clip.
[0039] Scalability is a desirable property for heterogeneous and
error prone environments. This property is desirable in order to
counter limitations such as constraints on bit rate, display
resolution, network throughput, and decoder complexity.
[0040] Scalability can be used to improve error resilience in a
transport system where layered coding is combined with transport
prioritisation. The term transport prioritisation here refers to
various mechanisms to provide different qualities of service in
transport, including unequal error protection, to provide different
channels having different error/loss rates. Depending on their
nature, data are assigned differently, for example, the base layer
may be delivered through a channel with high degree of error
protection, and the enhancement layers may be transmitted through
more error-prone channels.
[0041] Generally, scalable multimedia coding suffers from a worse
compression efficiency than non-scalable coding. In other words, a
multimedia clip coded as a scalable multimedia clip with all
enhancement layers requires greater bandwidth than if it had been
coded as a non-scalable single-layer clip with equal quality.
However, exceptions to this general rule exist, for example the
temporally scalable B-frames in video compression.
[0042] In the following, scalability is discussed with reference to
the ITU-T H.263 video compression standard. H.263 is an ITU-T
recommendation for video coding in low bit rate communication which
generally means data rates below 64 kbps. The recommendation
specifies the bitstream syntax and the decoding of the bitstream.
Currently, there are two versions of H.263. Version 1 consists of
the core algorithm and four optional coding modes. H.263 version 2
is an extension of version 1 providing twelve new negotiable coding
modes.
[0043] Pictures are coded as luminance and two colour difference
(chrominance) components (Y, C.sub.B and C.sub.R). The chrominance
pictures are sampled with a half of the pixels along the both
co-ordinate axes when compared to the luminance picture.
[0044] The scalability mode (Annex O) of H.263 specifies syntax to
support temporal, signal-to noise ratio (SNR), and spatial
scalability capabilities.
[0045] Spatial scalability and SNR scalability are closely related,
the only difference being the increased spatial resolution provided
by spatial scalability. An example of SNR scalable pictures is
shown in FIG. 6. SNR scalability implies the creation of multi-rate
bit streams. It allows for the recovery of coding errors, or
differences between an original picture and its reconstruction.
This is achieved by using a finer quantizer to encode the
difference picture in an enhancement layer. This additional
information increases the SNR of the overall reproduced
picture.
[0046] Spatial scalability allows for the creation of
multi-resolution bit streams to meet varying display requirements
and/or constraints. A spatially scalable structure is illustrated
in FIG. 7. It is essentially the same as in SNR scalability except
that a spatial enhancement layer attempts to recover the coding
loss between an up-sampled version of the reconstructed reference
layer picture and a higher resolution version of the original
picture. For example, if the reference layer has a quarter common
intermediate format (QCIF) resolution, and the enhancement layer
has a common intermediate format (CIF) resolution, the reference
layer picture must be scaled accordingly such that the enhancement
layer picture can be predicted from it. The QCIF standard allows
the resolution to be increased by a factor of two in the vertical
direction only, horizontal direction only, or both the vertical and
horizontal directions for a single enhancement layer. There can be
multiple enhancement layers, each increasing the picture resolution
over that of the previous layer. The interpolation filters used to
up-sample the reference layer picture are explicitly defined in the
H.263 standard. Aside from the up-sampling process from the
reference to the enhancement layer, the processing and syntax of a
spatially scaled picture are identical to those of an SNR scaled
picture.
[0047] In either SNR or spatial scalability, the enhancement layer
pictures are referred to as EI- or EP-pictures. If the enhancement
layer picture is upwardly predicted from a picture in the reference
layer, then the enhancement layer picture is referred to as an
Enhancement-I (EI) picture. In this type of scalability, the
reference layer means the layer "below" the current enhancement
layer. In some cases, when reference layer pictures are poorly
predicted, over-coding of static parts of the picture can occur in
the enhancement layer, causing an unnecessarily excessive bit rate.
To avoid this problem, forward prediction is permitted in the
enhancement layer. A picture that can be predicted in the forward
direction from a previous enhancement layer picture or,
alternatively, upwardly predicted from the reference layer picture
is referred to as an Enhancement-P (EP) picture. Note that
computing the average of the upwardly and forwardly predicted
pictures can provide bi-directional prediction for EP-pictures. For
both EI- and EP-pictures, upward prediction from the reference
layer picture implies that no motion vectors are required. In the
case of forward prediction for EP-pictures, motion vectors are
required.
[0048] In multi-point and broadcast multimedia applications,
constraints on network throughput may not be foreseen at the time
of encoding. Thus, a scalable bitstream should be used. FIG. 8
shows an IP multicasting arrangement where each router can strip
the bitstream according to its capabilities. It shows a server S
providing a bitstream to a number of clients C. The bitstreams are
routed to the clients by routers R. In this example, the server is
providing a clip which can be scaled to at least three bit rates,
120 kbit/s, 60 kbit/s and 28 kbit/s.
[0049] If the client and server are connected via a normal uni-cast
connection, the server may try to adjust the bit rate of the
transmitted multimedia clip according to the temporary channel
throughput. One solution is to use a layered bit stream and to
adapt to bandwidth changes by varying the number of transmitted
enhancement layers.
SUMMARY OF THE INVENTION
[0050] According to a first aspect of the invention there is
provided a method of transmitting a data signal having a sequence
of data units in a predetermined order over a transmission link
between a data source and a data sink comprising the step of
sending the data units in an order determined by their relative
importance rather than their predetermined order.
[0051] According to a second aspect of the invention there is
provided a data transmission system for transmitting a data signal
having a sequence of data units in a predetermined order over a
transmission link between a data source and a data sink the data
units being sent in an order determined by their relative
importance rather than their predetermined order.
[0052] According to a third aspect of the invention there is
provided a server for transmitting a data signal having a sequence
of data units in a predetermined order over a transmission link the
data units being sent in an order determined by their relative
importance rather than their predetermined order.
[0053] Preferably the server comprises re-ordering means.
Preferably the server comprises an editor for providing a scalable
data signal.
[0054] According to a fourth aspect of the invention there is
provided a computer program product stored on a computer usable
medium comprising computer readable program means for causing
transmission of a data signal having a sequence of data units in a
predetermined order over a transmission link the data units being
sent in an order determined by their relative importance rather
than their predetermined order.
[0055] Preferably the computer program product comprises a server.
In this sense it is meant that the when the computer program is
loaded into hardware it causes the hardware to operate as a server
device. Preferably the computer program product comprises an editor
for providing a scalable data signal. Preferably the computer
program product comprises re-ordering means for re-ordering the
data units.
[0056] According to a fifth aspect of the invention there is
provided a data signal having a sequence of data units for
transmission over a transmission link between a data source and a
data sink the data units being in an order determined by their
relative importance rather than their predetermined order.
[0057] Preferably where the data units or layers of the data units
are ordered into a sequence other than their original sequence, the
invention also includes means or a step for returning the data
units or layers of the data units into their original sequence.
[0058] The transmission link may simply be a bottleneck where
congestion is occurring and the invention is used to overcome the
bottleneck.
[0059] An advantage of the invention is that it can provide larger
safety times than prior art arrangements with the same initial
buffering. In addition, it provides some degree of protection
against decreased transmission throughput. Therefore it helps in
reducing the occurrence of pauses without sacrificing compression
efficiency.
[0060] The invention also helps in transmitting over a unreliable
network connection using transmission protocols which do not
automatically retransmit but only retransmit those missing parts
chosen by a higher level function, for example in response to
feedback. In this case, it provides more time for retransmission of
the vital parts of a data signal than prior art arrangements. Thus,
vital parts of a stream are more likely to be transmitted
successfully and the ultimate end user of the data signal is likely
to perceive a better output quality.
[0061] Preferably the bandwidth of the transmission link available
to the data signal is variable. This may be because the amount of
data traffic using the link is variable, for example due to the
need to re-transmit lost or corrupted data.
[0062] The invention provides a way of altering a data signal so
that it can be transmitted over a transmission link having a
bandwidth which would otherwise not be sufficient for it or for
allowing certain of the bandwidth of its transmission link to be
made available for other data, for example control information.
[0063] Preferably the effect of the invention is to increase the
safety times of some data units relative to, or even at the expense
of, the safety times of other data units.
[0064] Preferably the data signal is arranged in a hierarchy having
a base layer and at least one enhancement layer. The term base
layer means a most fundamental layer. The base layer may represent
the least refined version of the data signal. In each unit, the (or
each) enhancement layer, when combined with the base layer, refines
the base layer to make it more like the data unit. Each enhancement
layer may add more refinement to the base layer. Preferably the
safety time of the base layer is increased relative to the safety
times of the enhancement layers.
[0065] Preferably the data signal is scalable. The base layer and
the or each enhancement layer may have a scaled relationship with
respect to each other. The signal may be scalable in the temporal,
the spatial, the spectral or the SNR domains.
[0066] Preferably by virtue of re-ordering the sequence the layers
have safety times which depend on their position in the hierarchy.
For example the base layer of a particular data unit may have a
greater safety time than enhancement layers. Advantageously, the
finer the level of refinement represented by a layer, the shorter
its safety time.
[0067] Preferably the source is a server. Alternatively it is an
editor.
[0068] Preferably the sink is a client. It may be a destination of
the data signal such as a mobile terminal, for example a mobile
telephone.
[0069] Preferably the data units are frames. In one embodiment the
data units represent a sequence of pictures. Conveniently it is a
video sequence. Most preferably, it comprises multimedia data.
[0070] It should be understood that the terms "layer" and "layers"
are being used figuratively and refer to blocks of data which can
be applied to each other rather than to a strictly positional
relationship.
BRIEF DESCRIPTION OF THE DRAWINGS
[0071] The invention will now be described by way of example only
with reference to the accompanying drawings in which:
[0072] FIG. 1 shows a multimedia content creation and retrieval
system;
[0073] FIG. 2 shows operations of a multimedia clip editor;
[0074] FIG. 3 shows operations of a multimedia server;
[0075] FIG. 4 shows operations of a multimedia retrieval
client;
[0076] FIG. 5 shows an illustration of B picture prediction;
[0077] FIG. 6 shows an illustration of SNR scalability;
[0078] FIG. 7 shows an illustration of spatial scalability;
[0079] FIG. 8 shows scalable multimedia being used in an IP
multicast system;
[0080] FIG. 9 shows mobile multimedia retrieval system according to
the invention;
[0081] FIG. 10 shows operations of an editor according to the
invention;
[0082] FIG. 11 shows operations of a client according to the
invention;
[0083] FIG. 12 shows safety times of a re-ordered sequence and an
original sequence;
[0084] FIG. 13 shows receiving and playback of a re-ordered
sequence;
[0085] FIG. 14 also shows receiving and playback of a re-ordered
sequence; and
[0086] FIG. 15 shows another multimedia content creation and
retrieval system.
DETAILED DESCRIPTION
[0087] FIGS. 1 to 8 have been discussed above.
[0088] A mobile multimedia retrieval system 10 according to the
invention is shown in FIG. 9. Although only a single client 22 is
shown it will be appreciated that there can be, and in preferred
embodiments will be, a number of such clients. It will also be
appreciated that at least part of the transmission link connecting
a server 18 and the client 22 will be wireless.
[0089] The system 10 comprises an editor 12 which produces a
compressed clip, a server database 14 for storing the compressed
clip, re-ordering means 16 for re-ordering the sequence of the
compressed clip, a server 18 for sending the re-ordered compressed
clip and a client comprising order recovery means 20 to recover the
original sequence of the compressed clip. The client 22 controls
the operation of the server, for example by specifying which data
are sent and how they are sent. Clearly, the re-ordering means 16
and the order recovery means 20 need to be compatible so that the
latter can recover the original order. The re-ordering means 16
could be located to operate on data before they reach the server
database 14 rather than the embodiment shown in FIG. 9. Certain
types of data, such as a live stream, may by-pass the database 14
and be sent by the editor 12 directly to the re-ordering means 16.
Although they are shown separately in FIG. 9, the editor 12, the
server database 14, the re-ordering means 16 and the server 18 can
all be configured into a single unit or entity which can also be
referred to as a server.
[0090] FIG. 10 shows operation of an editor according to the
invention. Raw data captured in the form of one or more
uncompressed streams are edited to produce one or more uncompressed
tracks. These tracks are compressed in a compression step to
produce compressed tracks. The compressed tracks are multiplexed
together to produce a compressed multimedia clip. The multimedia
clips are coded scalably. The resulting bitstream consists of basic
units, such as frames in video compression. These basic units are
not all of equal importance; some are primary basic units which are
vital to decode the bitstream and some are secondary basic units
which improve the quality of the bitstream. Elimination of the
secondary basic units causes only a quality loss in the decoded
multimedia clip. In effect, the primary basic units are a base
layer of the bitstream and the secondary units are enhancement
layers.
[0091] Instead of organising the basic units into the original
transmission order, that is the sequence in which they are produced
by editing and compression of raw data, the invented method
re-orders their sequence by prioritising the primary basic units so
that they pre-empt the secondary basic units in the transmission
order. Reordering can take place before or after the multiplexing
process depending on the nature of the system and the data. In FIG.
10 it is shown as taking place after multiplexing.
[0092] A re-ordering algorithm is used to recorder the sequence. In
a preferred embodiment, the algorithm conducts stream analysis to
identify primary and secondary basic units and to collect bit rate
statistics. These may include the number of bits per frame. The
bitstream is then re-ordered with iterative re-ordering steps to
minimise the initial buffering time needed for full-quality
playback with full channel throughput and reduced-quality playback
with a reduced channel throughput. An example of such a re-ordering
algorithm is given below. It should be noted that the expected
reduced channel throughput has to be estimated in advance since it
is one of the input parameters of the algorithm. The estimation can
be based on the statistics obtained from the channel behaviour, for
example.
[0093] Re-ordering of the basic units means that the primary basic
units are likely to be received earlier than indicated in their
playback time even though some decrease in the channel throughput
occurs. However, secondary basic units are vulnerable to
transmission delay variations since the method schedules them to be
received just before their indicated playback time. In this way the
client is still likely to be able to play the primary basic units
without pausing the playback, even if the channel throughput
decreases considerably. The secondary basic units, on the other
hand, may arrive later than their scheduled playback time and will
not be used in playback.
[0094] FIG. 11 shows operation of a client according to the
invention. The bitstream is received by the client, re-ordered to
produce the original compressed clip, demultiplexed to produce the
compressed tracks and then decompressed to reconstruct the tracks.
The basic unit order recovery step in the client can be applied
either before or after demultiplexing (depending in when it occurs
in the content editor). In FIG. 11 it is shown as occurring before
demultiplexing, to correspond to the order described in FIG. 10. In
this way re-ordering and order recovery are both applied to the
multiplexed bitstream. Order recovery organises the basic units
back into their natural sequence for decompression and discards all
late-arriving secondary basic units.
[0095] On receipt of a basic unit by the client, it is time-stamped
with a time by which it should be played. This time-stamp is
compared with the current playback time. If the time-stamp is
greater than the current playback time the basic unit is not
decompressed and it is discarded. If the time-stamp is less than
the current playback time the basic unit is decompressed and the
two times are compared after decompression. If the time stamp is
less than the current playback time the basic unit is played.
Otherwise, it is discarded.
[0096] In the particular case in which re-ordering is performed
after multiplexing, the multiplexing/demultiplexing format used
needs to allow re-ordering of the basic units and recovery of the
original order.
[0097] FIG. 15 shows another view of a multimedia content creation
and retrieval system according to the invention. The system 50
comprises an editor 52, a server 54 and a client 56.
[0098] The editor comprises a compressor 58 for compressing a raw
data stream typically being in the form of a sequence of images
into a form suitable for transmission, a multiplexer 60 for
multiplexing the compressed raw data stream with other compressed
data streams together to produce a compressed bitstream and layer
generation means 62 for producing primary basic frames which are
vital to decode the bitstream and secondary basic frames which
improve the quality of the bitstream.
[0099] The server 54 comprises re-ordering means 64 for arranging
the frames in an order which depends on their importance in helping
to reconstruct the raw data stream at the client 56 rather than
their natural sequential order.
[0100] The server 54 and the client 56 each comprise
transmitting/receiving means 66 and 70 respectively for data
transmission over a transmission link 68 between the server 54 and
the client 56.
[0101] The client 56 comprises order recovery means 72 for putting
the frames back into their natural sequential order, a
demultiplexer 74 for demultiplexing the bitstream and a
decompressor 76 for decompressing the compressed data streams to
reproduce the raw data stream. The client comprises playing/output
means 78 for playing or otherwise outputting the raw data stream.
The client also comprises a controller 80 which applies a control
signal to the server 54 to control the re-ordering means in
response to receipt of frames in the client 56.
EXAMPLE
[0102] An embodiment of the invention was tested using an H.263
compliant encoder and a frame re-ordering tool. The encoder was
capable of producing B-frames. The re-ordering tool comprised a
re-ordering algorithm and could be adjusted to produce frame
ordering for different decreased channel data rates.
[0103] A standard ITU-T test sequence of video frames called
Glasgow was used. The sequence had a QCIF resolution, that is a
luminance picture size of 176.times.144 pixels and a chrominance
picture size of 88.times.72 pixels.
[0104] The first 297 frames of the Glasgow sequence were compressed
to produce a temporally scalable bitstream comprising INTRA (I)
frames, INTER (P) frames and bi-directionally predicted (B) frames.
The I and P frames are referred to as primary frames and the B
frames are referred to as secondary frames. The frame rate of the
original (default) sequence was 12.5 frames per second (fps) and so
the compressed sequence corresponded to 23.68 seconds. Each frame
was compressed. The first frame was an INTRA frame, and the second
one was an INTER frame. After that, there were alternate INTER
frames and B-frames. A constant quantizer was used to produce a
bitstream having an average bit rate of 41073 bps consisting of an
average bit rate for the primary frames of 28984 bps and an average
bit rate for the secondary frames of 12089 bps.
[0105] For the sake of simplicity, the example does not include
multiplexing of the bitstream with another bitstream.
[0106] Re-ordering of the frame sequence was optimised separately
for bit rates corresponding to sums of the primary frame bit rate
(A) and (i) 25%; (ii) 50%; and (iii) 75% of the secondary frame bit
rate (B). It should be noted that 100% of the secondary bit rate
(B) corresponds to the average bit rate required in order to obtain
full quality playback. The summed bit rates were 32007, 35029 and
38051 bps respectively. For convenience, in the following
description, these bit rates are referred to as 25% (A+0.25B), 50%
(A+0.5B) and 75% (A+0.75B), and the corresponding re-ordered
sequences are referred to as 25-ordered, 50-ordered and 75-ordered.
It was expected that the full rate of the channel would be the
average bit rate of the sequence.
[0107] The reordering algorithm used two values:
FQFR (full quality full rate), which is the minimum initial
buffering time to ensure full-quality playback assuming a full rate
channel; and
[0108] RQRR (reduced quality reduced rate) which is the minimum
initial buffering time to ensure reduced-quality playback in which
the primary frames are played and some or all secondary frames may
not be played (assuming a reduced secondary bit rate of 25%, 50% or
75%).
[0109] In calculating FQFR and RQRR for a particular channel, the
algorithm assumes that it has a constant bit rate. Using the bit
rate statistics the likely number of bits per frame is calculated.
The time instant at which a particular frame is to be played is
known from the time stamp. A time is determined by which a frame
will be ready for display as a picture (allowing, of course, for
the time needed to receive the bits for a frame and to prepare
them) and this time is compared with the time stamp of the frame.
If the frame will be received after it is to be displayed, the
buffering time is increased, if it is received in time then the
next frame can be dealt with. Once the algorithm has compared the
likely times of receipt of all of the frames against their playback
times and adjusted to buffering time accordingly so that they
should all be received in time, the minimum buffering time has been
calculated.
[0110] The algorithm was applied to a sequence having its original
order. In this case, FQFR is smaller than RQRR because in order to
obtain full quality playback, you have to wait longer in starting
playback if you have a reduced rate channel. The algorithm seeks to
minimise the initial buffering time needed for full-quality
playback with full channel throughput and at the same time for
reduced-quality playback with a reduced channel throughput. In
other words, the algorithm seeks to find a value where FQFR and
RQRR are approximately equal. It should be noted that the algorithm
is not optimal, that is the algorithm does not necessarily re-order
the sequence so that the minimum initial buffering delay is found.
The algorithm consists of the following steps:
1. Calculate FQFR and RQRR.
2. If FQFR is greater than or equal to RQRR, stop re-ordering. The
minimum initial buffering delay is the greater of FQFR or RQRR.
Otherwise, continue with the next step.
3. Shift all primary frames towards the beginning of the sequence
by one frame step. For example, if the initial sequence is,
I0, P1, P2, B3, P4, B5, P6, B7, P8, B9, P10, . . . ,
shifting changes the sequence to:
I0, P1, P2, P4, B3, P6, B5, P8, B7, P10, B9,
where I0 is the first frame. It can be seen that where a secondary
frame is earlier in sequence than an adjacent primary frame, the
positions of the primary and secondary frames are exchanged.
4. Continue from step 1.
[0111] The buffering time needed to ensure pause-less playback in
the reduced bit rate channel is greater for the original sequence
than for the re-ordered sequence. It is desirable to have a short
initial buffering time. The re-ordering process makes a trade-off
between the minimum buffering time needed for a full-rate channel
and the minimum buffering time needed for expected reduced-rate
channels. Without re-ordering, less initial buffering time would
have been needed to provide full-quality playback in the full-rate
channel. However, without re-ordering, more initial buffering time
would have been required to provide reduced-quality playback in the
reduced-rate channel.
[0112] In the example, safety times were also investigated. Initial
buffering time was set to be the minimum for reduced quality
playback of the re-ordered sequence at 25% secondary bit rate. FIG.
12 shows how safety time varies with playback of the 25-ordered
sequence and the original order sequence. It shows that throughout
playback, the 25-ordered sequence (represented by the upper line)
has a longer safety time than the original order sequence.
[0113] Results were also obtained for the 50% and 75% secondary bit
rates. The average safety times (in seconds) for all primary frames
in a sequence are shown in the following table for all three
reduced bit rates. The safety times are characterised by minimum
safety time and average safety time. TABLE-US-00001 25% 50% 75%
Minimum Re-ordered 4.79 2.51 1.06 Default 3.46 1.63 0.64 Difference
1.33 0.88 0.42 Average Re-ordered 5.90 3.51 2.12 Default 4.66 2.83
1.84 Difference 1.24 0.67 0.28
[0114] It can be seen that both the minimum and average safety
times of the re-ordered sequences are longer than those of the
original sequence.
[0115] In contrast, set out below is a table showing the required
buffering delays in seconds for full-quality playback at different
average bit rates. This is to minimise pauses in playback.
TABLE-US-00002 25% 50% 75% Re-ordered 10.04 6.03 3.53 Default 7.02
4.64 3.02
[0116] It can be seen that the re-ordered sequence needs a longer
buffering delay for full-quality playback because the secondary
frames are transmitted out of their natural order and time is
required to put all the frames back into their natural order.
However, this is not important because the invention sets out to
provide reduced quality playback with minimal initial buffering
delay. To illustrate this, the following table shows the buffering
delays required to ensure pause-less reduced quality playback:
TABLE-US-00003 25% 50% 75% Re-ordered 5.28 3.49 2.47 Default 6.91
4.46 2.84
[0117] These results show that where the bit rate of the
transmission channel is reduced, the invented method requires a
shorter buffering delay before playback can be started.
[0118] The effect of the invention is illustrated in FIG. 13 which
shows playback of the 25-ordered sequence at 25% bit rate (that is
A+0.25B as discussed above). Playback times, that is the times at
which the frames are intended or supposed to be played to a viewer,
are represented as a straight line. To be played in sequence,
without requiring pauses, each frame must be received before its
playback time. Any symbol located below the playback time line
represents a frame which is received before its scheduled playback
time and any symbol located above the playback time line represents
a frame which is received after its scheduled playback time. In the
example of FIG. 13, the primary frames were all received in time to
be played and the secondary frames were all received too late.
Therefore the secondary frames were all discarded without being
played.
[0119] It should be understood that if the sequence had not been
re-ordered, there is a likelihood that some of the primary frames
would have arrived after their playback times and so would have
required playback to be paused (since a primary frame cannot be
discarded without also discarding all of the frames predicted from
it).
[0120] FIG. 14 shows how the 25-ordered sequence is received and
played when the channel throughput is not reduced. It can be seen
that all frames are received in time to be played. Again, the upper
line represents when secondary frames are received and the lower
line represents when primary frames are received.
[0121] Of course, the exact appearance of FIGS. 13 and 14 depends
on the nature of the original sequence, as well as the bit rate and
the re-ordering that is performed. Differences can be expected in
other systems which re-order data in different ways.
[0122] In use, the invention can be applied on-line to prepare a
bitstream for transmission at some future point. For example,
assuming that there may be considerable variation in the available
bandwidth of a given transmission channel, the algorithm may be
applied to a bitstream and re-ordered sequences may be calculated
for several alternative bandwidths, such as 25%, 50% and 75%. In
transmission, if problems are encountered, for example with
pausing, the re-ordered sequence currently being used may be
changed to one better suited to a lower available bandwidth.
Equally, if transmission is proceeding better than expected, a
sequence better suited for a higher bandwidth may be used. The
invention may check the progress of transmission and/or playback a
number of times during transmission and adjust the sequence
accordingly. It is advantageous if this adjustment can be
programmed dynamically whilst transmission is occurring. In this
way, intermittent problems can de dealt with.
[0123] The part of the invention which re-orders a bitstream is
located in an editor or in a server. The part of the invention
which recovers the order is located in a client.
[0124] A further improvement is provided if the proportion of bits
belonging to the primary frames is reduced by replacing some of the
primary frames with B-frames. This reduces the compression
efficiency slightly because B-frames are predicted from reference
frames (P-frames) which are further away in time and so the
B-frames and P-frames are less alike. This yields a poorer
predicted B-frame and so more bits must be used to code the
corresponding prediction error image. Since in this arrangement
there are now more bits belonging to secondary frames which can be
discarded, re-ordering the frames can provide even better
protection against transmission delays and pauses.
[0125] One way of looking at the invention is to say that it
sacrifices the safety times of the secondary frames in order to
maintain maximum safety times for the primary frames.
[0126] The invention is particularly suitable for streaming-type
multimedia retrieval.
[0127] While preferred embodiments of the invention have been shown
and described, it will be understood that such embodiments are
described by way of example only. For example, although the
invention has been described applied to a temporally scalable
bitstream, it can also be applied to other types of scalability,
including spatial, spectral and SNR. Numerous variations, changes
and substitutions will occur to those skilled in the art without
departing from the scope of the present invention. Accordingly, it
is intended that the following claims cover all such variations as
fall within the spirit and scope of the invention.
* * * * *