System and method for using audio performance to control...

Multiplex communications – Communication techniques for information carried in plural... – Adaptive

Reexamination Certificate

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Reexamination Certificate

active

06373855

ABSTRACT:

FIELD OF THE INVENTION
The field of the invention relates to the transmission of multimedia information, and in particular to using the audio performance of a channel to control the video bandwidth of a signal with audio and video components.
BACKGROUND OF THE INVENTION
Audio-video (A-V) signals that are sent over a channel from an A-V sender to an A-V recipient are used to convey audio and video information for applications such as videotelephone conversations, television entertainment programming, and live news and sporting events. A channel is an entity that carries information between a sender and a recipient at a finite (although possible variable) bandwidth. “Bandwidth” means rate of information. For example, bandwidth is measured in bits per second. In another example, bandwidth is measured in characters per second.
In many A-V applications, receiving audio information above a certain minimum level of quality is more important than receiving the accompanying video information. For example, in videotelephony, it can be more important to receive an understandable audio signal than 30 frames-per-second of video information.
The channel over which A-V information is conveyed has a finite overall bandwidth, i.e., capacity to carry information. This finite bandwidth must be shared by the audio portion and the video portion of an A-V signal. In many cases, the overall bandwidth is not large enough to carry an A-V signal at the highest quality with which it can be sent. For example, an A-V sender can generate an A-V signal where the audio portion is music-quality at 40,000 bits per second, and the video portion is full-motion video at 30 frames per second at 120,000 bits per second. However, the overall bandwidth of the channel is only 80,000 bits per second. At this time, the sender must decide how much of the 80,000 bits per second of overall bandwidth available on the channel to allocate to the audio signal, and how much to allocate to the video signal. For example, the A-V sender may decide to send the audio signal at 20,000 bits per second, and the video signal at only 15 frames per second at 60,000 bits per second. Together, the audio and video signals in this example fill the 80,000 bits per second of overall bandwidth available on the channel.
Known techniques adapt the bandwidth of a signal to be sent based upon performance parameters of, for example, the processor performance of the recipient. For example, in one known system, a file is transferred from a sender to a recipient in blocks. Initially, the blocks are sent at a default size. If fewer than a given number of bad blocks are reported by the recipient for the most recent blocks attempted to be transferred, then the block size is increased. If at any time during the transfer at least a certain number of bad blocks are reported for the most recent blocks attempted to be transferred, then the sender reduces the block size. An example of an adaptive transfer protocol is described in
Receiver-driven Lavered Multicast
, by Steven McCanne, Van Jacobson and Martin Vetterli, ACM SIGCOMM '96, Aug. 1996, Stanford Calif. However, such adaptive file transfer protocols do not distinguish between different types of content being transferred, and do not adjust the rate at which any one kind of content is transferred.
Other known A-V systems adapt the bandwidth of an A-V signal based upon information received from the recipient pertaining to the usage of the recipient processor. This is meant to prevent the recipient processor from becoming overwhelmed by having to process too rapid a rate of incoming information from the sender. One such known system is used by the ProShare® application made by the Intel Corporation of Santa Clara, Calif. The ProShare® application is a videotelephone call that uses a microphone, speaker display and camera at both sender and recipient computers. ProShare® uses an algorithm that changes the bandwidth of an A-V signal. This algorithm operates independently of any signal received from a sender. Rather, when executed, the ProShare® application starts streaming local video from the recipient's own camera to a video window on the recipient's display. ProShare® monitors the load on the recipient's processor, and aims at using about 75-80% of the recipient's processor at all times. If another application is executed using the recipient's processor that consumes x % of the processor's capacity, then the algorithm in ProShare® immediately restricts the bandwidth of the A-V signal to the extent necessary to reduce its load on the processor by about 75-x %. This effectively reduces the bandwidth of the generated video signal. Likewise, the bandwidth of the A-V signal is increased when more processor capacity becomes available. The same algorithm operates in processing an A-V signal from a sender during a videotelephone call. Similarly, the video bandwidth is reduced if the sender detects congestion on the outgoing network channel. This occurs if the bit rate of the signal sent from the sender exceeds the bandwidth of the transport medium (e.g., a connection between sender and receiver that implements packet framing and channel allocation, Ethernet, etc.) A standard for packet framing and channel allocation is described in International Telecommunication Union Standard ITU-T H.221. A standard for Ethernet is described in Institute for Electrical and Electronics Engineers (IEEE) Standard 802.3.
In another known system, the audio portion of an A-V signal is allocated a fixed (guaranteed) amount of bandwidth over a channel, and the rest of the bandwidth is shared by other information, e.g., video, data, etc. One such known system uses a 2 KB-128KB Basic Rate Integrated Services Digital Network (ISDN) connection using International Telecommunication Union Standard ITU-T H.221, where ProShare® uses the ITU-T H.320 protocol described in International Telecommunication Union Standard ITU-T Recommendation H.320. Standards describing ISDN include Bellcore Standards TR444, TR448, TR317, TR394 and TR268. Other standards describing ISDN are published by the ISDN.ISO 8877, published by the International Telecommunication Union. In this known system, the audio portion of a signal is allocated a fixed (guaranteed) amount of bandwidth. If, for example, a whiteboard application is executed during a call and whiteboard application data is streamed between sender and recipient, ProShare® tries to allocate enough bandwidth for the whiteboard data. Whatever is left of the 128K bandwidth slice is allocated to video. The allocation of bandwidth is thus somewhat predefined in the H.320 standard. Because the audio bandwidth is almost guaranteed, it is not necessary to have the bandwidth of the video signal adjusted to allow for adequate bandwidth for the audio signal.
Known systems that adjust the rate at which information flows between sender and recipient based upon processor utilization ignore the problem of dynamically and efficiently responding to changes in the quality of the link between a sender and a recipient. For example, a processor may be free to accept a high bandwidth A-V signal, but the quality of the link may be such that only a small bandwidth of A-V information may be reliably transmitted. Known techniques provide no way of preferentially allocating audio bandwidth over video bandwidth in response to link quality.
Known systems that use fixed bandwidth allocations also ignore link quality and can be inefficient. An audio bandwidth of a fixed size can be inefficiently large for a link of high quality, and inadequate for a link of low quality.
A better system is needed that can dynamically and preferentially allocate A-V bandwidth based upon link quality.
SUMMARY OF THE INVENTION
An embodiment of the present invention provides a method for allocating video bandwidth of a channel that carries both video and audio information. In accordance with an embodiment of the present invention, the quality of an audio signal received over the channel is determined. The video bandwidth of th

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