System and method for improving audio quality on a...

Multiplex communications – Special services – Conferencing

Reexamination Certificate

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Details

C704S500000

Reexamination Certificate

active

06240070

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention generally relates to teleconferencing, and in particular, to a system and method that improves audio quality by compensating for lossy audio compression algorithms used in conventional conferencing networks.
2. Description of the Related Art
Computer-based conferencing has become popular as a means for carrying out a meeting between remotely located participants. As shown in
FIG. 1
, a conventional conferencing network
10
typically includes a plurality of clients
14
-
16
connected to a conference server
12
via a communications network
15
. The server
12
permits the clients
14
-
16
to simultaneously communicate with one another, effectively emulating a conference room filled with participants.
Although many computer-based conferencing systems permit transfer of video, voice, and data, the system
10
shown in
FIG. 1
illustrates only the audio portion of the conferencing system. In the system
10
, the clients
14
-
16
can be any device permitting a user to transmit and receive compressed audio data, such as a personal computer (PC), a video phone, a telephone featuring audio compression, or the like. Each of the clients
14
-
16
includes an audio compression algorithm
18
-
20
and a corresponding decompression algorithm
22
-
24
. When a participant speaks, his/her utterances are converted to digital information by the respective clients
14
-
16
. This digital information is then compressed by the compression algorithms
18
-
20
and transferred over the network
15
to the conference server
12
.
To maintain a fully duplexed audio channel, the conference server
12
separately decompresses each of the compressed audio streams received from the clients
14
-
16
using the decompressor
26
. The decompressed audio channels are then summed by a summation circuit
28
. The output of the summation circuit is then compressed by a compressor
30
. Representing the fully duplexed channel, the compressed summation stream is then transmitted back to each of the clients
14
-
16
over the network
15
. Each of the clients
14
-
16
includes a decompression algorithm
22
-
24
which decompresses the audio information carried in the common channel.
In recent years, there has been major progress in the development of compression algorithms. This represents a real economic benefit, since compressed voice takes much less bandwidth than uncompressed voice and many conversations can be multiplexed over the same channel. However, compression algorithms such as GSM, True Speech™, G.723.1, etc., are lossy and hence, pose a problem for features like conference calling. Using lossy algorithms results in data loss, and consequently introduces distortism every time the audio is compressed. This problem becomes more compounded in hierarchical networks, where summation streams from various servers are successively combined using lossy compression algorithms. Every round of compression/decompression results in a poorer voice quality.
Alternatively, lossless compression algorithms can be used to overcome the problems caused by repeated transformations. However, lossless algorithms have lower compression ratios than lossy algorithms and therefore consume much more bandwidth. In addition, lossy compression is essential in narrow band communication systems, such as systems using H.324.
Therefore, there is a need for an improved conferencing system that does not rely on lossless audio algorithms and compensates for degraded audio quality caused by lossy compression transformations.
SUMMARY OF THE INVENTION
It is an advantage of the present invention to provide an improved conferencing system and method that reduces the negative effect of lossy algorithms on audio quality.
In accordance with one embodiment of the present invention, there is provided a system that includes an improved conference server. Audio quality is improved by selectively allowing a dominant input channel to bypass a lossy compression transform performed by the server.
The conference server includes an input interface that receives compressed audio channels from a plurality of clients using lossy compression. Coupled to the interface is a decompressor, which decompresses the incoming audio channels to produce corresponding decompressed audio channels. A summation circuit sums the decompressed audio channels to produce a summation stream. The summation stream is then recompressed and provided as a first output of the server.
A selection unit included in the server selects a dominant channel from among the incoming compressed audio channels. The audio information carried in the dominant channel is then directly provided as a second output of the server, bypassing the decompression/compression round which normally takes place in a conventional server. By allowing one or more incoming audio channels to selectively bypass the lossy transformation of the server, the overall quality of audio presented at the clients can be improved. Thus, the audio information of the second output leaves the server having a significantly higher audio quality.
At each client, various mixing options are available to the user. For instance, the compressed dominant channel of the second output can be combined with the compressed summation stream of the first output to produce overall higher quality conference audio.


REFERENCES:
patent: 5383184 (1995-01-01), Champion
patent: 5434913 (1995-07-01), Tung et al.
patent: 5570363 (1996-10-01), Holm
patent: 5983192 (1999-11-01), Botzko et al.
patent: 6064653 (2000-05-01), Farris
patent: 6070089 (2000-05-01), Brophy et al.

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