Abstract:
A decoder for decompressing real-time media data streams and a method for operating such a decoder is disclosed. The decoder may comprise a relatively larger first memory for storing compressed data and parameters, and a processor for executing a decompression algorithm, the processor having a relatively smaller second memory. The decompression algorithm may be executed on a periodic basis, and the parameters used to select the data to be decompressed may be moved from the second memory to the first memory each time the decompression algorithm executes. An embodiment of the present invention may use slower, less expensive memory to enable it to support a greater number of real-time media streams than prior art solutions. Another embodiment of the present invention may include machine-readable storage having stored thereon a computer program having a plurality of code sections executable by a machine for causing the machine to perform the foregoing.

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS/INCORPORATION BY REFERENCE 
   [Not Applicable.] 
   FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT 
   [Not Applicable] 
   [MICROFICHE/COPYRIGHT REFERENCE] 
   [Not Applicable] 
   BACKGROUND OF THE INVENTION 
   Network traffic for real-time media consists of one or more streams of data packets, each stream supporting one “channel”. Each packet of the stream provides a limited amount of playback time for the associated channel. In order to provide continuous playback, the data packets for each channel must arrive at regular intervals. The time that a packet takes to traverse the network varies, however, due to a number of factors. These factors include, for example, the number of nodes, the speed of the communications links, and the queuing delay that occurred at each node in the path. In addition, data packets may be lost in transit. Packet loss and variations in network delay, normally referred to as ‘network delay jitter’, occur as a part of normal packet network operation. Minimizing the effects of network delay jitter and packet loss on the playback of a real-time media stream is a challenging problem, and involves the buffering of the compressed real-time data at the point of playback for each channel. The buffers used for this purpose are referred to as “jitter buffers”. 
   Each packet in the data stream for a channel includes a small amount of packet header information, and a much larger amount of compressed real-time data, or “payload.” Typically, the packet header and the payload are stored together in the jitter buffer for that channel. Although the information in the packet header is used repeatedly by the jitter buffer algorithms, the payload is accessed only when decompression occurs. 
   The digital signal processors typically used to implement the decompression algorithms have a relatively small amount of fast, internal data memory. When a typical jitter buffer is stored within the internal data memory of the digital signal processor, it occupies a large portion of that space. Each channel that is supported also requires a certain amount of memory space during decompression for program or “instance” variables. The combined memory requirement of the jitter buffer and the instance variables limits the number of channels that may be supported by the internal memory of the typical digital signal processor. As an alternative, the jitter buffer may be stored in a larger external memory. Storing the jitter buffer in external memory, however, slows access to the packet header information needed for the jitter buffer algorithms. The slower access to packet header information reduces processor throughput, limiting the number of channels that may be supported. 
   Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with some aspects of the present invention as set forth in the remainder of the present application with reference to the drawings. 
   BRIEF SUMMARY OF THE INVENTION 
   Aspects of the present invention may be seen in a decoder comprising a processor for executing at least one decompression algorithm, the processor comprising a first memory, and a second memory for storing at least one of the payload data, header data, and algorithm instance data. At least one of the header data, the payload data, and the algorithm instance data may be moved from the second memory to the first memory just prior to the execution of the at least one decompression algorithm, and at least one of the header data and the algorithm instance data may be moved from the first memory to the second memory following the execution of the at least one decompression algorithm. The payload data may comprise speech data, and the header data may comprise at least one of a time stamp, a sequence number, a jitter estimate, and a reference to a location within the second memory. 
   In an embodiment of the present invention, the size of the first memory may be a small fraction of the size of the second memory. In addition, the first memory may have a relatively higher speed of access and the second memory may have a relatively lower speed of access. The at least one decompression algorithm may be executed on a periodic basis. 
   Another aspect of the present invention may be observed in a method of operating a decoder for decoding a real-time media stream, the method comprising receiving a plurality of data packets where each of the plurality of data packets may comprise header data and payload data, and parsing each of the plurality of data packets into a header data portion and a payload data portion. The method may further comprise storing the header data portions in a first memory, thereby forming a block of header data, and storing the payload data portions in the first memory, thereby forming a block of payload data. The method may also comprise copying the block of header data to a second memory, decoding at least a portion of the block of payload data using the copy of the block of header data in the second memory, and moving at least a portion of the copy of the block of header data in the second memory back to the first memory. 
   In an embodiment in accordance with the present invention, the payload data may comprise compressed speech data, and the header data may comprise at least one of a time stamp, a sequence number, a jitter estimate, and a reference to a location within the first memory. In addition, the second memory and a digital signal processor may be contained within a single integrated circuit device. The size of the second memory may be a small fraction of the size of the first memory, and the copying, decoding, and moving may occur on a periodic basis. 
   Yet another aspect of the present invention may be observed in a machine-readable storage, having stored thereon a computer program having a plurality of code sections for implementing a decoder, the code sections executable by a machine for causing the machine to perform the foregoing. 
   These and other advantages, aspects, and novel features of the present invention, as well as details of illustrated embodiments, thereof, will be more fully understood from the following description and drawings. 

   
     BRIEF DESCRIPTION OF SEVERAL VIEWS OF THE DRAWINGS 
       FIG. 1  is a functional block diagram representing a communication system that enables the transmission of real-time media data over a packet-based system. 
       FIG. 1A  is a functional block diagram representing another communication system that enables the transmission of real-time media data over a packet-based system. 
       FIG. 2  is a block diagram of an exemplary embodiment illustrating the services invoked by a packet voice transceiver system, in accordance with the present invention. 
       FIG. 3  is a more detailed block diagram showing the network services invoked by the network VHD operating in the voice mode and the associated PXD. 
       FIG. 4  shows a block diagram illustrating the storage arrangement of a jitter buffer in which the header data and payload data have been stored in separate memory areas having different speed of access, in accordance with the present invention. 
       FIG. 5  shows a block diagram of an exemplary decoder that may correspond, for example, to the decoder of  FIG. 2 , in accordance with the present invention. 
       FIG. 6  is a flow chart illustrating a method of operating a decoder, in accordance with the present invention. 
   

   DETAILED DESCRIPTION OF THE INVENTION 
   The invention described relates in general to the processing of a payload media stream. More specifically, the present invention pertains to the processing of multiple, compressed, real-time media streams in a system with limited high-speed memory. 
   In an illustrative embodiment of the present invention, a signal processing system is employed to interface voice telephony devices with packet-based networks. Voice telephony devices include, by way of example, analog and digital phones, Ethernet phones, IP phones, interactive voice response systems, private branch exchanges (PBXs) and any other conventional voice telephony devices known in the art. The described embodiment of the signal processing system can be implemented with a variety of technologies including, by way of example, embedded communications software that enables transmission of voice data over packet-based networks. The embedded communications software may be run on programmable digital signal processors (DSPs), and used in gateways, remote access servers, PBXs, and other packet-based network appliances. Although the embodiments described below are with respect to the use of the invention(s) within systems performing voice communication, the embodiments described herein are for illustrative purposes only, as the present invention is not limited in this respect and may have significant utility in systems used for the communication of other real-time media, for example, voice, music, video, etc. 
   Referring now to  FIG. 1 , there is shown a functional block diagram representing a communication system that enables the transmission of voice data over a packet-based system such as voice-over-IP (VoIP, H.323), Voice over Frame Relay (VoFR, FRF-11), Voice Telephony over ATM (VTOA), or any other proprietary network, according to an illustrative embodiment of the present invention. In one embodiment of the present invention, voice data can also be carried over traditional media such as time division multiplex (TDM) networks and voice storage and playback systems. Packet-based network  10  provides a communication medium between telephony devices. Network gateways  12   a  and  12   b  support the exchange of voice between packet-based network  10  and telephony devices  13   a  and  13   b . Network gateways  12   a  and  12   b  may include a signal processing system that provides an interface between the packet-based network  10  and telephony devices  13   a  and  13   b . Network gateway  12   c  supports the exchange of voice between packet-based network  10  and a traditional circuit-switched network  19 , which transmits voice data between packet-based network  10  and telephony device  13   c . In the described exemplary embodiment, each network gateway  12   a ,  12   b ,  12   c  supports a telephony device  13   a ,  13   b ,  13   c.    
   Each network gateway  12   a ,  12   b ,  12   c  could support a variety of different telephony arrangements. By way of example, each network gateway might support any number of telephony devices, circuit-switched networks and/or packet-based networks including, among others, analog telephones, Ethernet phones, fax machines, data modems, PSTN lines (Public Switched Telephone Network), ISDN lines (Integrated Services Digital Network), T1 systems, PBXs, key systems, or any other conventional telephony device and/or circuit-switched/packet-based network. In the described exemplary embodiment, two of the network gateways  12   a ,  12   b  provide a direct interface between their respective telephony devices and the packet-based network  10 . The other network gateway  12   c  is connected to its respective telephony device through a circuit-switched network such as a PSTN  19 . The network gateways  12   a ,  12   b ,  12   c  permit voice, fax and modem data to be carried over packet-based networks such as PCs running through a USB (Universal Serial Bus) or an asynchronous serial interface, Local Area Networks (LAN) such as Ethernet, Wide Area Networks (WAN) such as Internet Protocol (IP), Frame Relay (FR), Asynchronous Transfer Mode (ATM), Public Digital Cellular Network such as TDMA (IS-13x), CDMA (IS-9x), or GSM for terrestrial wireless applications, or any other packet-based system. 
   Another exemplary topology is shown in  FIG. 1A . The topology of  FIG. 1A  is similar to that of  FIG. 1  but includes a second packet-based network  16  that is connected to packet-based network  10  and to telephony device  13   b  via network gateway  12   b . The signal processing system of network gateway  12   b  provides an interface between packet-based network  10  and packet-based network  16  in addition to an interface between packet-based networks  10 ,  16  and telephony device  13   b . Network gateway  12   d  includes a signal processing system that provides an interface between packet-based network  16  and telephony device  13   d.    
   Referring now to  FIG. 2 , there is illustrated a signal flow diagram of a packet voice transceiver system  200 , in accordance with an embodiment of the present invention. In an illustrative embodiment of the present invention, the packet voice transceiver system  200  may reside in a network gateway such as network gateways  12   a ,  12   b ,  12   c  of  FIG. 1 , and  12   a ,  12   b ,  12   c , and  12   d  of  FIG. 1A . In an exemplary embodiment, packet voice transceiver system  200  provides two-way communication with a telephone or a circuit-switched network, such as a PSTN line (e.g. DS 0 ). The packet voice transceiver  200  includes a Virtual Hausware Driver (VHD)  205 , a switchboard  210 , a physical device driver (PXD)  215 , an interpolator  220 , and a decimator  225 . 
   The VHD  205  is a logical interface to a telephony device such as  13   a ,  13   b , and  13   c  of  FIG. 1 , via the packet network  10 , and performs functions such as voice encoding and decoding, media queue management, dual tone multi-frequency (DTMF) detection and generation, and call discrimination (CDIS). During a communication session (e.g., voice, video, fax) each telephony device associates a VHD  205  with each of the telephony device(s) with which it is communicating. For example, during a voice-over-packet (VoIP) network call between telephony devices  13   a  and  13   b , telephony device  13   a  associates a VHD  205  with telephony device  13   b , and telephony device  13   b  associates a VHD  205  with telephony device  13   a . Communication between telephony devices  13   a  and  13   b  takes place through their respective VHD 205 , and packet network  10 . 
   The switchboard  210  associates the VHD  205  and the PXD  215  engaged in a communication session by supporting the connection and combination of data streams from the VHD 205  and PXD 215  assigned to the telephony devices participating in the session. 
   The PXD  215  represents an interface for transmitting and receiving the input and output signals to and from the user, and performs various functions including, for example, echo cancellation. As shown in  FIG. 2 , the top of the PXD  215  interfaces with switchboard  210 , while the bottom of the PXD  215  passes data to the interpolator  220  and receives data from decimator  225 . The functions within a wideband PXD  215  may be designed to use, for example, 16 kHz sampled data, while functions in a narrowband PXD  215  may expect to process, for example, 8 kHz sampled data. 
   A wideband system may contain a mix of narrowband and wideband VHDs  205  and PXDs  215 . A difference between narrowband and wideband device drivers is their ingress and egress sample buffer interface. A wideband VHD  205  or PXD  215  has wideband data at its sample buffer interface and includes wideband services and functions. A narrowband VHD  205  or PXD  215  has narrowband data at its sample buffer interface and can include narrowband services and functions. The switchboard interfaces with narrowband and wideband VHDs  205  and PXDs  215  through their sample buffer interfaces. The switchboard  210  is incognizant of the wideband or narrowband nature of the device drivers, but is aware of the sampling rate of the data that it reads and writes data through the sample buffer interfaces. To accommodate differences in the sampling rates of data streams, an embodiment of the present invention may upsample data received from narrowband sources and downsample data being sent to narrowband destinations. The sample buffer interfaces may provide data at any arbitrary sampling rate. In an embodiment of the present invention, the narrowband sample buffer interface may provide data sampled at 8 kHz and the wideband sample buffer interface may provide data sampled at 16 kHz. Additionally, a VHD  205  may be dynamically changed between wideband and narrowband and vice versa. 
   The VHD  205  and PXD  215  driver structures may include sample rate information to identify the sampling rates of the wideband and narrowband data. The information may be part of the interface structure that the switchboard understands and may contain a buffer pointer and an enumeration constant or the number of samples to indicate the sample rate. 
   The packet voice transceiver system  200  is also characterized by an ingress path and an egress path, in which the ingress path transmits user packets to a packet network such as, for example, packet network  10  of  FIG. 1 , and the egress path receives user packets from a packet network such as, for example, packet network  10  of  FIG. 1 . The ingress path and the egress path can either operate in a wideband support mode or a narrowband support mode, and the ingress path and the egress path are not required to operate in the same mode. For example, the ingress path can operate in the wideband support mode, while the egress path operates in the narrowband mode. 
   In the exemplary embodiment shown in  FIG. 2 , the ingress path comprises the decimator  225 , echo canceller  235 , switchboard  210 , and services including but not limited to DTMF detector  240  and CDIS  245 , and packet voice engine (PVE)  255  comprising an encoder algorithm  260 , and packetization function  261 . In the ingress path of a wideband device, the decimator  225  receives the user inputs and provides, for example, 16 kHz sampled data for an 8 kHz band-limited signal. The 16 kHz sampled data is transmitted through echo canceller  235  and switchboard  210  to the VHD  205  associated with the destination telephony device. In some cases, the DTMF detector  240  may be designed for operation on only narrowband digitized samples, and the wideband data may be downsampled and passed to DTMF detector  240 . Similarly, where CDIS  245  is designed for operation on only narrowband digitized samples, downsampled wideband data may be provided to CDIS  245 , which distinguishes a voice call from a facsimile transmission. 
   The PVE  255  is responsible for issuing media queue mode change commands consistent with the active voice encoder and decoder. The media queues can comprise, for example, the media queues described in patent application Ser. No. 10/313,826, “Method and System for an Adaptive Multimode Media Queue”, filed Dec. 6, 2002, which is incorporated herein by reference in its entirety. The PVE  255  ingress thread receives raw samples from other functions within VHD  205 . Depending upon the operating mode of VHD  205 , the raw samples include either narrowband or wideband data. At PVE  255 , encoder  260  encodes and packetizes the sampled data into compressed speech frames for transmission over a packet network such as, for example, packet network  10  of  FIG. 1 . The encoder  260  can comprise, for example, the BroadVoice  32  Encoder made by Broadcom, Inc. 
   The egress path comprises depacketizer  262 , decoder  263 , CDIS  266 , DTMF generator  269 , switchboard  210 , echo canceller  235 , and interpolator  220 . The depacketizer  262  receives data packets from a packet network such as, for example packet network  10  of  FIG. 1 , passing the compressed speech frames to the decoder  263 . The decoder  263  can comprise, for example, the BroadVoice  32  decoder made by Broadcom, Inc. The decoder  263  decodes the compressed speech frames received from the depacketizer  262  and may provide wideband sampled data. If CDIS  266  and DTMF generator support 16 kHz sampled data, the 16 kHz sampled is provided to CDIS  266  and DTMF generator  269 . Again, in one embodiment, where CDIS  266  and DTMF generator  269  require narrowband digitized samples, the wideband data may be downsampled and used by CDIS  266  and the DTMF generator  269 . 
   The DTMF generator  269  generates DTMF tones if detected in the data packets received from the sending telephony device  13   a ,  13   b , and  13   c . These tones may be written to the wideband data to be passed to switchboard  210 . The wideband data is received by the switchboard  210 , which provides the data to the PXD  215 . The sampled data is passed through the echo canceller  235  and provided to interpolator  220 . 
     FIG. 3  is a more detailed block diagram showing the network services invoked by the network VHD  62  in the voice mode and the associated PXD  60 . In the described exemplary embodiment, the PXD  60  provides two-way communication with a telephone or a circuit-switched network, such as a PSTN line (e.g. DS 0 ) carrying a 64 kb/s pulse code modulated (PCM) signal, i.e., digital voice samples. 
   The incoming PCM signal  60   a  is initially processed by the PXD  60  to remove far-end echoes that might otherwise be transmitted back to the far-end user. As the name implies, echoes in telephone systems are the return of the talker&#39;s voice resulting from the operation of the hybrid with its two-four wire conversion. If there is low end-to-end delay, echo from the far end is equivalent to side-tone (echo from the near-end), and therefore, not a problem. Side-tone gives users feedback as to how loudly they are talking, and indeed, without side-tone, users tend to talk too loudly. However, far-end echo delays of more than about 10 to 30 msec significantly degrade the voice quality and are a major annoyance to the user. 
   An echo canceller  70  is used to remove echoes from far-end speech present on the incoming PCM signal  60   a  before routing the incoming PCM signal  60   a  back to the far-end user. The echo canceller  70  samples an outgoing PCM signal  60   b  from the far-end user, filters it, and combines it with the incoming PCM signal  60   a . Preferably, the echo canceller  70  is followed by a non-linear processor (NLP)  72  which may mute the digital voice samples when far-end speech is detected in the absence of near-end speech. The echo canceller  70  may also inject comfort noise which in the absence of near-end speech may be roughly at the same level as the true background noise or at a fixed level. 
   After echo cancellation, the power level of the digital voice samples is normalized by an automatic gain control (AGC)  74  to ensure that the conversation is of an acceptable loudness. Alternatively, the AGC can be performed before the echo canceller  70 . However, this approach would entail a more complex design because the gain would also have to be applied to the sampled outgoing PCM signal  60   b . In the described exemplary embodiment, the AGC  74  is designed to adapt slowly, although it should adapt fairly quickly if overflow or clipping is detected. The AGC adaptation should be held fixed if the NLP  72  is activated. 
   After AGC, the digital voice samples are placed in the media queue  66  in the network VHD  62  via the switchboard  32 ′. In the voice mode, the network VHD  62  invokes three services, namely call discrimination, packet voice exchange, and packet tone exchange. The call discriminator  68  analyzes the digital voice samples from the media queue to determine whether a 2100 Hz tone, a 1100 Hz tone or V.21 modulated HDLC flags are present. If either tone or HDLC flags are detected, the voice mode services are terminated and the appropriate service for fax or modem operation is initiated. In the absence of a 2100 Hz tone, a 1100 Hz tone, or HDLC flags, the digital voice samples are coupled to the encoder system which includes a voice encoder  82 , a voice activity detector (VAD)  80 , a comfort noise estimator  81 , a DTMF detector  76 , a call progress tone detector  77  and a packetization engine  78 . 
   Typical telephone conversations have as much as sixty percent silence or inactive content. Therefore, high bandwidth gains can be realized if digital voice samples are suppressed during these periods. A VAD  80 , operating under the packet voice exchange, is used to accomplish this function. The VAD  80  attempts to detect digital voice samples that do not contain active speech. During periods of inactive speech, the comfort noise estimator  81  couples silence identifier (SID) packets to a packetization engine  78 . The SID packets contain voice parameters that allow the reconstruction of the background noise at the far end. 
   From a system point of view, the VAD  80  may be sensitive to the change in the NLP  72 . For example, when the NLP  72  is activated, the VAD  80  may immediately declare that voice is inactive. In that instance, the VAD  80  may have problems tracking the true background noise level. If the echo canceller  70  generates comfort noise during periods of inactive speech, it may have a different spectral characteristic from the true background noise. The VAD  80  may detect a change in noise character when the NLP  72  is activated (or deactivated) and declare the comfort noise as active speech. For these reasons, the VAD  80  should generally be disabled when the NLP  72  is activated. This is accomplished by a “NLP on” message  72   a  passed from the NLP  72  to the VAD  80 . 
   The voice encoder  82 , operating under the packet voice exchange, can be a straight 16-bit PCM encoder or any voice encoder which supports one or more of the standards promulgated by ITU. The encoded digital voice samples are formatted into a voice packet (or packets) by the packetization engine  78 . These voice packets are formatted according to an applications protocol and sent to the host (not shown). The voice encoder  82  is invoked only when digital voice samples with speech are detected by the VAD  80 . Since the packetization interval may be a multiple of an encoding interval, both the VAD  80  and the packetization engine  78  should cooperate to decide whether or not the voice encoder  82  is invoked. For example, if the packetization interval is 10 msec and the encoder interval is 5 msec (a frame of digital voice samples is 5 ms), then a frame containing active speech should cause the subsequent frame to be placed in the 10 ms packet regardless of the VAD state during that subsequent frame. This interaction can be accomplished by the VAD  80  passing an “active” flag  80   a  to the packetization engine  78 , and the packetization engine  78  controlling whether or not the voice encoder  82  is invoked. 
   In the described exemplary embodiment, the VAD  80  is applied after the AGC  74 . This approach provides optimal flexibility because both the VAD  80  and the voice encoder  82  are integrated into some speech compression schemes such as those promulgated in ITU Recommendations G.729 with Annex B VAD (March 1996)—Coding of Speech at 8 kbits/s Using Conjugate-Structure Algebraic-Code-Exited Linear Prediction (CS-ACELP), and G.723.1 with Annex A VAD (March 1996)—Dual Rate Coder for Multimedia Communications Transmitting at 5.3 and 6.3 kbit/s, the contents of which is hereby incorporated herein by reference as though set forth in full herein. 
   Operating under the packet tone exchange, a DTMF detector  76  determines whether or not there is a DTMF signal present at the near end. The DTMF detector  76  also provides a pre-detection flag  76   a  which indicates whether or not it is likely that the digital voice sample might be a portion of a DTMF signal. If so, the pre-detection flag  76   a  is relayed to the packetization engine  78  instructing it to begin holding voice packets. If the DTMF detector  76  ultimately detects a DTMF signal, the voice packets are discarded, and the DTMF signal is coupled to the packetization engine  78 . Otherwise the voice packets are ultimately released from the packetization engine  78  to the host (not shown). The benefit of this method is that there is only a temporary impact on voice packet delay when a DTMF signal is pre-detected in error, and not a constant buffering delay. Whether voice packets are held while the pre-detection flag  76   a  is active could be adaptively controlled by the user application layer. 
   Similarly, a call progress tone detector  77  also operates under the packet tone exchange to determine whether a precise signaling tone is present at the near end. Call progress tones are those which indicate what is happening to dialed phone calls. Conditions like busy line, ringing called party, bad number, and others each have distinctive tone frequencies and cadences assigned them. The call progress tone detector  77  monitors the call progress state, and forwards a call progress tone signal to the packetization engine to be packetized and transmitted across the packet based network. The call progress tone detector may also provide information regarding the near end hook status which is relevant to the signal processing tasks. If the hook status is on hook, the VAD should preferably mark all frames as inactive, DTMF detection should be disabled, and SID packets should only be transferred if they are required to keep the connection alive. 
   The decoding system of the network VHD  62  essentially performs the inverse operation of the encoding system. The decoding system of the network VHD  62  comprises a de-packetizing engine  84 , a voice queue  86 , a DTMF queue  88 , a precision tone queue  87 , a voice synchronizer  90 , a DTMF synchronizer  102 , a precision tone synchronizer  103 , a voice decoder  96 , a VAD  98 , a comfort noise estimator  100 , a comfort noise generator  92 , a lost packet recovery engine  94 , a tone generator  104 , and a precision tone generator  105 . 
   The de-packetizing engine  84  identifies the type of packets received from the host (i.e., voice packet, DTMF packet, call progress tone packet, SID packet), transforms them into frames which are protocol independent. The de-packetizing engine  84  then transfers the voice frames (or voice parameters in the case of SID packets) into the voice queue  86 , transfers the DTMF frames into the DTMF queue  88  and transfers the call progress tones into the call progress tone queue  87 . In this manner, the remaining tasks are, by and large, protocol independent. 
   A jitter buffer is utilized to compensate for network impairments such as delay jitter caused by packets not arriving with the same relative timing in which they were transmitted. In addition, the jitter buffer compensates for lost packets that occur on occasion when the network is heavily congested. In the described exemplary embodiment, the jitter buffer for voice includes a voice synchronizer  90  that operates in conjunction with a voice queue  86  to provide an isochronous stream of voice frames to the voice decoder  96 . 
   Sequence numbers embedded into the voice packets at the far end can be used to detect lost packets, packets arriving out of order, and short silence periods. The voice synchronizer  90  can analyze the sequence numbers, enabling the comfort noise generator  92  during short silence periods and performing voice frame repeats via the lost packet recovery engine  94  when voice packets are lost. SID packets can also be used as an indicator of silent periods causing the voice synchronizer  90  to enable the comfort noise generator  92 . Otherwise, during far-end active speech, the voice synchronizer  90  couples voice frames from the voice queue  86  in an isochronous stream to the voice decoder  96 . The voice decoder  96  decodes the voice frames into digital voice samples suitable for transmission on a circuit switched network, such as a 64 kb/s PCM signal for a PSTN line. The output of the voice decoder  96  (or the comfort noise generator  92  or lost packet recovery engine  94  if enabled) is written into a media queue  106  for transmission to the PXD  60 . 
   The comfort noise generator  92  provides background noise to the near-end user during silent periods. If the protocol supports SID packets, (and these are supported for VTOA, FRF-11, and VoIP), the comfort noise estimator at the far-end encoding system should transmit SID packets. Then, the background noise can be reconstructed by the near-end comfort noise generator  92  from the voice parameters in the SID packets buffered in the voice queue  86 . However, for some protocols, namely, FRF-11, the SID packets are optional, and other far-end users may not support SID packets at all. In these systems, the voice synchronizer  90  continues to operate properly. In the absence of SID packets, the voice parameters of the background noise at the far end can be determined by running the VAD  98  at the voice decoder  96  in series with a comfort noise estimator  100 . 
   Preferably, the voice synchronizer  90  is not dependent upon sequence numbers embedded in the voice packet. The voice synchronizer  90  can invoke a number of mechanisms to compensate for delay jitter in these systems. For example, the voice synchronizer  90  can assume that the voice queue  86  is in an underflow condition due to excess jitter and perform packet repeats by enabling the lost frame recovery engine  94 . Alternatively, the VAD  98  at the voice decoder  96  can be used to estimate whether or not the underflow of the voice queue  86  was due to the onset of a silence period or due to packet loss. In this instance, the spectrum and/or the energy of the digital voice samples can be estimated and the result  98   a  fed back to the voice synchronizer  90 . The voice synchronizer  90  can then invoke the lost packet recovery engine  94  during voice packet losses and the comfort noise generator  92  during silent periods. 
   When DTMF packets arrive, they are de-packetized by the de-packetizing engine  84 . DTMF frames at the output of the de-packetizing engine  84  are written into the DTMF queue  88 . The DTMF synchronizer  102  couples the DTMF frames from the DTMF queue  88  to the tone generator  104 . Much like the voice synchronizer, the DTMF synchronizer  102  is employed to provide an isochronous stream of DTMF frames to the tone generator  104 . Generally speaking, when DTMF packets are being transferred, voice frames should be suppressed. To some extent, this is protocol dependent. However, the capability to flush the voice queue  86  to ensure that the voice frames do not interfere with DTMF generation is desirable. Essentially, old voice frames which may be queued are discarded when DTMF packets arrive. This will ensure that there is a significant gap before DTMF tones are generated. This is achieved by a “tone present” message  88   a  passed between the DTMF queue and the voice synchronizer  90 . 
   The tone generator  104  converts the DTMF signals into a DTMF tone suitable for a standard digital or analog telephone. The tone generator  104  overwrites the media queue  106  to prevent leakage through the voice path and to ensure that the DTMF tones are not too noisy. 
   There is also a possibility that DTMF tone may be fed back as an echo into the DTMF detector  76 . To prevent false detection, the DTMF detector  76  can be disabled entirely (or disabled only for the digit being generated) during DTMF tone generation. This is achieved by a “tone on” message  104   a  passed between the tone generator  104  and the DTMF detector  76 . Alternatively, the NLP  72  can be activated while generating DTMF tones. 
   When call progress tone packets arrive, they are de-packetized by the de-packetizing engine  84 . Call progress tone frames at the output of the de-packetizing engine  84  are written into the call progress tone queue  87 . The call progress tone synchronizer  103  couples the call progress tone frames from the call progress tone queue  87  to a call progress tone generator  105 . Much like the DTMF synchronizer, the call progress tone synchronizer  103  is employed to provide an isochronous stream of call progress tone frames to the call progress tone generator  105 . And much like the DTMF tone generator, when call progress tone packets are being transferred, voice frames should be suppressed. To some extent, this is protocol dependent. However, the capability to flush the voice queue  86  to ensure that the voice frames do not interfere with call progress tone generation is desirable. Essentially, old voice frames which may be queued are discarded when call progress tone packets arrive to ensure that there is a significant inter-digit gap before call progress tones are generated. This is achieved by a “tone present” message  87   a  passed between the call progress tone queue  87  and the voice synchronizer  90 . 
   The call progress tone generator  105  converts the call progress tone signals into a call progress tone suitable for a standard digital or analog telephone. The call progress tone generator  105  overwrites the media queue  106  to prevent leakage through the voice path and to ensure that the call progress tones are not too noisy. 
   The outgoing PCM signal in the media queue  106  is coupled to the PXD  60  via the switchboard  32 ′. The outgoing PCM signal is coupled to an amplifier  108  before being outputted on the PCM output line  60   b.    
   Referring for a moment to  FIG. 2 , the functionality of the VHD  205  is responsible for processing the egress packet stream or “voice channel” received from each of the far-end packet voice transceiver systems  200  engaged in a communication session. For example, in a call involving three participants using telephony devices  13   a ,  13   b , and  13   d  of  FIG. 1A , the packet voice transceiver  200  associated with each telephony device may process two voice channels using two VHD  205   s , one for each of the two other telephony devices. The jitter buffer within the decoder  263  of each VHD  205  compensates for irregularities in the arrival of voice packets from the associated far-end packet voice transceiver  200 , by storing speech data sufficient to bridge delays in packet arrival. The amount of memory needed for the jitter buffer in decoder  263  depends upon a number of factors including but not limited to the expected network delay jitter, and the rate at which the contents is consumed by playback. In the case of the packet voice transceiver  200  of  FIG. 2 , an amount of data equivalent to 300 milliseconds (ms) of speech playback may need to be buffered to avoid audible impairments due to network delay jitter. Depending upon network conditions, a greater or lesser amount of memory may be needed. In addition, the actual amount of jitter buffer space needed for the storage of 300 ms of speech data varies based upon the algorithm used to encode the speech. For example, speech encoded using the International Telecommunications Union—Telecommunications Standards Sector (ITU-T) G.711 standard may require 1200 16-bit words for the storage of 300 ms worth of compressed speech data, and an additional 300 16-bit words for the storage of the associated packet headers. In contrast a voice coder such as the BV32 encoder by Broadcom, Inc. may require only 600 16-bit words for speech data storage, half that of the G.711 standard. 
     FIG. 4  shows a block diagram illustrating the storage arrangement of a jitter buffer  400  in which the header data and payload data have been stored in separate memory areas having different speed of access, in accordance with the present invention. The jitter buffer  400  may correspond, for example, to the jitter buffer used by the decoder  263  of  FIG. 2 , or by the voice queue  86  of  FIG. 3 . As described above, a decoder such as the decoder  263  of  FIG. 2  may use a jitter buffer  400  in the processing of each voice channel. In the exemplary embodiment of  FIG. 4 , the memory space allocated for jitter buffer  400  has been partitioned into two segments, a header memory  406  and a payload memory  407 . In the illustration of  FIG. 4 , four voice packets have been separated into a header data portion and a payload data portion. The header data portion of each of the four packets is stored in header memory  406  as header data  410 ,  430 ,  450 , and  470 , while the payload data portion of each of the four packets is stored in payload memory  407  as payload data  420 ,  440 ,  460 , and  480 , respectively. For ease of understanding, the illustration of  FIG. 4  shows the header data and payload data portions corresponding to only four voice packets. An embodiment of the present invention is not limited in this manner, and may be adapted for use with the header data and payload data from a greater or lesser number of packets, without departing from the spirit of the invention. 
   The algorithms used in the jitter buffer of decoder  263  of  FIG. 2  may include, for example, the tracking of network delay jitter, the detection of packets that are received out of order or lost, and the calculation of the time of release of the packets to the speech decoding algorithms. In performing these and other functions, the jitter buffer and decoder algorithms typically make frequent use of the information contained within the header data  410 ,  430 ,  450 , and  470 . In addition, algorithm “instance” data are heavily accessed during the operation of the jitter buffer and decoder algorithms. In order to maximize the throughput of those algorithms, an embodiment in accordance with the present invention may store header data  410 ,  430 ,  450 , and  470 , and algorithm instance data (not shown) in memory that allows the fastest possible access. 
   Although the header data  410 ,  430 ,  450 , and  470 , and algorithm instance data may be needed on a frequent basis, the payload data  420 ,  440 ,  460 , and  480  may be needed only when decoding of the speech data takes place. The payload data from each network packet represents speech playback of a limited duration, for example, 5 milliseconds. Depending upon the speed of the processor used in the implementation of the decoder  263 , the decoding of the speech data contained within payload data  420 ,  440 ,  460 , and  480  may take only a small fraction of the time of the actual speech playback. Actual playback of the speech data contained within the payload data  420 ,  440 ,  460 , and  480  may involve infrequent and limited access to memory, when compared to that for the header data  410 ,  430 ,  450 , and  470 . 
     FIG. 5  shows a block diagram of an exemplary decoder  500  that may correspond, for example, to the decoder  263  of  FIG. 2 , in accordance with the present invention. In the illustration of  FIG. 5 , decoder  500  comprises digital signal processor (DSP)  510 , depacketizer  540 , external memory  547 , and bus  545 . The DSP  510  is further comprised of central processing unit (CPU)  520  and random access memory (RAM)  530 . External memory  547  is partitioned into one jitter buffer for each channel of real-time data supported by the decoder  500 , in this case jitter buffer  550  and jitter buffer  560 . 
   In the exemplary decoder  500  of  FIG. 5 , depacketizer  540  receives packets from egress packet stream  505 . Egress packet stream  505  may correspond, for example, to a stream of packets from packet network  10  of  FIG. 1 . Depacketizer  540  disassembles each received packet and stores the packet contents into jitter buffer  550  or jitter buffer  560  for the associated speech channel. For example, the header data and the corresponding payload data from a received packet may be stored in jitter buffer  550  of external memory  547 , as one of header data  551 ,  553 ,  555 , and  557 , and one of payload data  552 ,  554 ,  556 , and  558 , respectively. In the exemplary embodiment shown in  FIG. 5 , the header data may include, for example, packet sequence numbers, time stamps, and jitter estimates, while the payload data may comprise, for example, compressed speech, music, or video data. 
   In the exemplary embodiment shown in  FIG. 5 , the RAM  530  is arranged to contain header data  531  and algorithm instance data  532 . The RAM  530  may reside on the same integrated circuit (IC) as the CPU  520 , allowing the CPU  520  to have the fastest possible access to the contents of the RAM  530 . Although the RAM  530  is shown as being connected only to the CPU  520 , the RAM  530  may be connected to bus  545  and operate in a dual-port fashion with depacketizer  540 , without departing from the spirit of the present invention. The RAM  530  may be capable of storing, for example, 32 kilobytes of data, and may be limited in size due to the cost of the chip area occupied by the RAM  530 . In an embodiment of the present invention, external memory  547  may be considerably larger that the RAM  530  and may be, for example, several megabytes in size. The speed of access to the external memory  547  by the CPU  520  may be considerably slower than the access to the RAM  530 . 
   An embodiment in accordance with the present invention may take advantage of the relatively high speed of the DSP  510  and RAM  530  by keeping the most frequently used information, the header data and the algorithm instance data, within the RAM  530  during the processing of the associated payload data. Due to the limited size of the RAM  530 , an embodiment of the present invention may copy or ‘page’ portions of the slower external memory  547  into the faster RAM  530 , before processing by the CPU  520 . As shown in the illustration of  FIG. 5 , in an embodiment in accordance with the present invention, the RAM  530  has been arranged with space for header data  531  and algorithm instance data  532 . The header data  551 ,  553 ,  555 , and  557  of jitter buffer  550 , or header data  561 ,  563 ,  565 ,  567  of jitter buffer  560  is stored within the RAM  530  during processing of the associated payload data by the decoder algorithms. The header data  551 ,  553 ,  555 , and  557 , and header data  561 ,  563 ,  565 , and  567  may correspond, for example, to header data  410 ,  430 ,  450 , and  470  of  FIG. 4 . The RAM  530  may also contain instance data  532 . Instance data  532  may correspond, for example, to instance data  559  or  569 , and may include, for example, intermediate values of calculations and other algorithm variables used during the decoding of payload data  552 ,  554 ,  556 , and  558 , and payload data  562 ,  564 ,  566 , and  568 , respectively. 
   In an embodiment in accordance with the present invention, the DSP  510  may periodically perform the decoding functions for multiple speech channels in a round-robin, channel-by-channel fashion. This is because the processing time needed to decode a predetermined amount of the payload data for a channel may be a small fraction of the playback time of the decoded data. For example, let us assume that the speech data for the next channel to be processed is stored in jitter buffer  550 . Immediately prior to processing the speech data for the current channel, the CPU  520  may copy the header data  551 ,  553 ,  555 , and  557 , and instance data  559  into the header data  531  portion and the instance data  532  portion of the RAM  530 , respectively. The amount of payload data to be processed may represent, for example, 5 ms. of speech playback time. The CPU  520  may then perform the decoding of that portion of payload data  552 ,  554 ,  556 , and  558  needed for the next interval of speech playback, according to the algorithm used for the current channel. The processing needed to decode 5 ms. of playback may take, for example, 700 microseconds. Upon completion of the processing, the CPU  520  may copy to the header data  551 ,  553 ,  555 , and  557  of jitter buffer  550 , the header data  531  corresponding to those portions of payload data  552 ,  554 ,  556 , and  558  that have not yet been processed. It may also copy the current instance data  532  to the instance data  559  of jitter buffer  550 . The CPU  520  may then copy the header data  561 ,  563 ,  565 , and  567 , and instance data  569  from the jitter buffer  60  to the header data  531  and instance data  532  portions of the RAM  530 , respectively, and execute the decoding algorithm for the channel associated with jitter buffer  560 . 
   By storing only the header and instance data for the jitter buffer of a channel in the RAM  530 , and storing them in the RAM  530  only during the actual processing of the corresponding payload data, it is possible for an embodiment(s) of the present invention to perform the decoding of a greater number of speech channels than if the entire jitter buffer for all channels were always stored in the RAM  530 . In addition, the CPU  520  may have, for example, several MIPS of computing capacity available to perform other signal processing including, but not limited to, for example, tone generation or detection, echo cancellation or suppression, comfort noise generation, or similar functions. The exact amount of processor capacity available depends upon, for example, the number and size of the jitter buffers implemented, and the types of vocoders in use. In an embodiment of the present invention, a fast DSP  510  with a limited amount of RAM  530  may use larger, less expensive external memory  520  to process a larger number of speech channels that prior art solutions. Although described with respect to the processing of speech channels, the present invention is not limited to its use in speech applications, and may have significant utility with other real-time media streams (e.g., music, video, etc.) 
     FIG. 6  is a flow chart illustrating a method of operating a decoder, in accordance with the present invention. In the flow diagram shown in  FIG. 6 , two branches are shown, representing two processes that may take place in parallel. The processes illustrated may be performed by one processor, or by a number of processors operating in cooperation, without departing from the spirit of the present invention. In the first branch of  FIG. 6 , a speech packet for voice channel ‘J’ is received (block  610 ), and the header data portion of the packet is stored in a part of a relatively slower, larger first memory (block  612 ) reserved for the header data of voice channel ‘J’. The payload data or compressed speech portion of the received packet is then stored in a separate part of the slower, larger memory (block  614 ) reserved for the payload data of voice channel ‘J’. Although the exemplary method of  FIG. 6  is described with respect to speech, the present invention is not limited in this regard, as the present invention is applicable to other real-time media as well. 
   In the second branch of  FIG. 6 , a decoder such as, for example, decoder  263  of  FIG. 2 , processes, in sequence, the payload data for each of the supported voice channels. It begins by copying to a relatively faster, smaller memory all header data and algorithm instance data corresponding to the current channel from a part of the slower, larger memory reserved for header data and algorithm instance data of the current voice channel (block  616 ). The decoder algorithm then processes the oldest payload data for the current voice channel that is stored in the faster, smaller memory (block  618 ), using the header data and algorithm instance data stored in the slower, larger memory. Upon completion, the decoder copies header data and algorithm instance data from the faster, smaller memory to the area of the slower, larger memory reserved for the header data and algorithm instance data for the current voice channel (block  620 ). A check is then made whether processing for all channels has been completed (block  622 ). If not all channels have been processed, the decoder copies to the faster, smaller memory all header data and algorithm instance data corresponding to the next voice channel from a part of the slower memory reserved for header data and algorithm instance data of the next voice channel (block  624 ), and the sequence continues until processing for all voice channels is completed. 
   Although the present invention has been described above primarily with respect to its application to voice communication systems, it is not limited in this regard. The present invention may also be applied to other real-time communication media as well, e.g. music, video, etc., without departing from its spirit or scope. 
   Accordingly, the present invention may be realized in hardware, software, or a combination of hardware and software. The present invention may be realized in a centralized fashion in one computer system, or in a distributed fashion where different elements are spread across several interconnected computer systems. Any kind of computer system or other apparatus adapted for carrying out the methods described herein is suited. A typical combination of hardware and software may be a general-purpose computer system with a computer program that, when being loaded and executed, controls the computer system such that it carries out the methods described herein. 
   The present invention also may be embedded in a computer program product, which comprises all the features enabling the implementation of the methods described herein, and which when loaded in a computer system is able to carry out these methods. Computer program in the present context means any expression, in any language, code or notation, of a set of instructions intended to cause a system having an information processing capability to perform a particular function either directly or after either or both of the following: a) conversion to another language, code or notation; b) reproduction in a different material form. 
   Notwithstanding, the invention and its inventive arrangements disclosed herein may be embodied in other forms without departing from the spirit or essential attributes thereof. Accordingly, reference should be made to the following claims, rather than to the foregoing specification, as indicating the scope of the invention. In this regard, the description above is intended by way of example only and is not intended to limit the present invention in any way, except as set forth in the following claims. 
   While the present invention has been described with reference to certain embodiments, it will be understood by those skilled in the art that various changes may be made and equivalents may be substituted without departing from the scope of the present invention. In addition, many modifications may be made to adapt a particular situation or material to the teachings of the present invention without departing from its scope. Therefore, it is intended that the present invention not be limited to the particular embodiment disclosed, but that the present invention will include all embodiments falling within the scope of the appended claims.