Multiplex communications – Pathfinding or routing – Switching a message which includes an address header
Reexamination Certificate
1999-11-15
2004-03-02
Ngo, Ricky (Department: 2697)
Multiplex communications
Pathfinding or routing
Switching a message which includes an address header
C370S252000, C714S748000
Reexamination Certificate
active
06700893
ABSTRACT:
TECHNICAL FIELD OF THE INVENTION
The present invention is directed, in general, to data streaming systems and, more specifically, to a decoder buffer for use in a streaming data receiver, such as a streaming video receiver.
BACKGROUND OF THE INVENTION
Real-time streaming of data, such as multimedia content, over Internet protocol (IP) networks has become an increasingly common application in recent years. A wide range of interactive and non-interactive multimedia Internet applications, such as news on-demand, live TV viewing, video conferencing, and many others, rely on end-to-end streaming solutions. Unlike a “downloaded” audio or video file, which may be retrieved first in “non-real” time and viewed or played back later, streaming audio and streaming video applications require an audio source or video source to encode and to transmit an audio signal or video signal over a network to an audio receiver or a video receiver, which must decode and display (or play back) the transmitted signal in real time. The receiver relies on a decoder buffer to receive encoded video data packets and/or encoded audio data packets from the network and to transfer the packets to a video decoder and/or an audio decoder.
Two problems arise when a streaming data signal is transmitted across a non-guaranteed Quality-of-Service (QoS) network, such as the Internet. First, end-to-end variations in the network (e.g., delay jitter) between the streaming data transmitter and the streaming data receiver mean that the end-to-end delay is not constant. Second, there is usually a significant packet loss rate across non-QoS networks, often requiring re-transmission. The lost data packet must be recovered prior to the time the corresponding frame must be decoded. If not, an underflow event occurs. Furthermore, if prediction-based compression is used, an underflow due to lost data packets may not only impact the current frame being processed, but may affect many subsequent frames.
It is well-known that re-transmission of lost packets is a viable means of recovery for continuous media communication over packet networks. Many applications use a negative automatic repeat request (NACK) in conjunction with re-transmission of the lost packet. These approaches take into consideration both the round-trip delay and the delay jitter between the sender and the receiver(s).
For example, an end-to-end model with re-transmission for packet voice transmission has been developed. This model takes advantage of the fact that voice data consists of periods of silence separated by brief talk-spurt segments. The model also assumes that each talk-spurt consists of a fixed number of fixed-size packets. However, this model is not general enough to capture the characteristics of compressed video or audio (which can have variable number of bytes or packets per video or audio frame). Additionally, an adaptive playback algorithm has been developed that changes the playback time of a video frame in response to network conditions. This results in a time-varying playback rate (i.e., introduces “playback jitter”) in response to network jitter and packet losses.
The above mentioned solutions can be applicable for voice data or for certain video applications which tolerate “playback jitter.” However, these solutions may not be acceptable for many types of video-on-demand services (e.g., entertainment applications). In addition, while maintaining continuous decoding and displaying of the real-time audio/visual data, it is crucial for the selected packet loss recovery mechanism to modify its operation according to changing conditions during the Internet session in which the data is transmitted.
Any packet retransmission scheme must strike a balance in determining when to request retransmission of a late data packet. If a streaming data receiver waits too long before requesting retransmission of a late (and possibly lost) data packet, the requested data packet may not be received when needed, due to the round trip delay associated with the retransmission request and retransmission of the late data packet. However, if the streaming data receiver waits only a very brief period before requesting retransmission of a late (but not lost) data packet, an excessive amount of the limited bandwidth available between the streaming data transmitter and the streaming data receiver will be consumed by the increased number of unnecessary retransmission requests and the increased number of duplicate packet transmissions.
There is therefore a need in the art for improved streaming data receivers that compensate for variations inherent in a non-QoS network. In particular, there is a need for an improved receiver decoder buffer that takes into consideration both transport delay parameters (e.g., end-to-end delay and delay jitter) and video. (or audio) encoder buffer constraints. More particularly, there is a need for an improved decoder buffer that implements a packet loss recovery mechanism that modifies its operation according to changing conditions of the data network over which the streaming data is transmitted and minimizes the number of duplicate data packet transmissions.
SUMMARY OF THE INVENTION
The present invention is embodied in an Integrated Transport Decoder (ITD) buffer model. One key advantage of the ITD model is that it eliminates the separation of a network-transport buffer, which is typically used for removing delay jitter and recovering lost data, from the video/audio decoder buffer. This can significantly reduce the end-to-end delay, and optimize the usage of receiver resources (such as memory).
The present invention provides a re-transmission framework that uses a time-delay budget constraint for streaming video receiver during a real-time Internet session. In other words, at the beginning of the session, the streaming data receiver introduces a certain start-up delay to the incoming bitstream. This start-up delay defines the time-delay budget that the streaming data receiver can rely on for packet loss recovery for the remainder of the session. The re-transmission framework manages this time-delay budget in an adaptive manner in response to changing network conditions. The present invention maximizes the time for uninterrupted decoding and presentation of the multimedia content while minimizing time for duplicate-packet transfer events. These duplicate-packet transfer events occur when the streaming data receiver requests the re-transmission of packets prematurely, reducing the effective available bandwidth between the streaming data transmitter and the streaming data receiver.
It is a primary object of the present invention to provide a delay budget controller for use with a decoder buffer capable of receiving streaming data packets over a data network from a streaming transmitter and storing the data packets in a plurality of access units for subsequent retrieval by a streaming data decoder. In an advantageous embodiment, the delay budget controller comprises 1) a first controller capable of monitoring at least one network parameter associated with the data network; and 2) a second controller capable of monitoring in the decoder buffer a delay budget region comprising a sequence of access units that are about to be accessed sequentially by the data decoder, the delay budget region comprising a retransmission region and a late region separated by a temporal boundary, wherein the second controller detects missing data packets in the retransmission region and the late region and, in response to detection of a missing data packet in the retransmission region, transmits a retransmission request for the missing data packet to the streaming transmitter, and wherein the second controller is capable of adjusting the temporal boundary to thereby advance or retard the transmission of the retransmission request.
In one embodiment of the present invention, the second controller adjusts the temporal boundary in response to a measured value of the at least one network parameter.
In another embodiment of the present invention, the at least one network param
Loguinov Dmitri
Radha Hayder
Gross Russell
Koninklijke Philips Electronics , N.V.
Ngo Ricky
Robustelli Michael E.
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