Voice and data exchange over a packet based network with resource management

Abstract

A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Claims

What is claimed is:

1. A method of managing resources of a system, comprising:

processing a signal;

estimating signal processing complexity; and

adjusting adaptation speed of an echo canceller for processing the signal, when the estimated complexity exceeds a threshold, wherein the signal processing comprises adaptively canceling the echos from the signal, and the estimating signal processing complexity comprises estimating echo return loss enhancement (ERLE) of the echo canceller.

2. The method of claim 1 further comprising bypassing the echo canceller and suppressing echo of the signal by an echo suppressor instead, when the estimated complexity exceeds a threshold.

3. The method of claim 1 wherein the estimating signal processing complexity comprises estimating maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

4. A method of managing resources of a system, comprising:

performing a plurality of signal processing functions on a signal, including echo cancellation function;

estimating average complexity of each of the signal processing functions;

summing the estimated average complexity of the each of the signal processing functions; and

adjusting adaptation speed of the echo cancellation function, when the sum of the estimated average complexities exceeds a threshold, wherein the estimating signal processing complexity comprises estimating maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

5. The method of claim 4 further comprising bypassing the echo cancellation function and suppressing echo of the signal by an echo suppressor instead, when the sum of the estimated average complexities exceeds a threshold.

6. The method of claim 4 wherein adjusting adaptation speed of the echo cancellation function comprises reducing the complexity of the echo cancellation adaption.

7. A data transmission system, comprising:

a telephony device which outputs a signal; and

a signal processor coupled to the telephony device, the signal processor comprising a resource manager that estimates signal processor complexity based on characteristics of the signal, and adjusts adaptation speed of an echo canceller for processing the signal by changing the number of coefficients of the echo canceller, when the estimated complexity exceeds a threshold, wherein the resource manager estimates signal processor complexity by estimating echo return loss enhancement (ERLE) of the echo canceller.

8. The data transmission system of claim 7 wherein the signal processor comprises an echo suppressor, and the resource manager reduces the signal processor complexity by bypassing the echo canceller and suppressing echo of the signal by the echo suppressor instead, when the estimated complexity exceeds a threshold.

9. The data transmission system of claim 7 wherein the resource manager estimates maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

10. A resource manager for a signal processor, comprising:

estimation means for estimating signal processor complexity based on characteristics of the signal; and

adjusting means for adjusting adaptation speed of an echo canceller for processing the signal by changing the number of coefficients of the echo canceller, when the estimated complexity exceeds a threshold, wherein the estimation means comprises means for estimating maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

11. The resource manager of claim 10 further comprising echo suppressing means, and wherein the adjusting means comprises means for bypassing the echo canceller and suppressing echo of the signal by the echo suppression means instead, when the estimated complexity exceeds a threshold.

12. A resource manager for a signal processor performing a plurality of functions including echo cancellation function, comprising:

estimation means for estimating average complexity of each of the functions by comparing a first signal to a second signal;

summing means for summing the estimated average complexity of each of the functions; and

adjusting means adjusting adaptation speed of the echo cancellation function by changing the number of coefficients of an echo canceller, when the sum of the estimated average complexities exceeds a threshold, wherein the adjusting means comprises means for estimating maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

13. The resource manager of claim 12 further comprising echo suppress means, and wherein the adjusting means comprises means for bypassing the echo canceller and suppressing echo of the signal by the echo suppressing means instead, when the estimated complexity exceeds a threshold.

14. Computer-readable media embodying a program of instructions executable by a computer to perform a method of managing resources of a signal processing system, the method comprising:

estimating signal processing complexity based on characteristics of the signal; and

adjusting adaptation speed of an echo canceller for processing the signal by changing the number of coefficients of the echo canceller, when the estimated complexity exceeds a threshold, wherein the estimating signal processing complexity comprises estimating maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

15. The computer-readable media of claim 14 further comprising instructions for bypassing the echo canceller and suppressing echo of the signal by an echo suppressor instead, when the estimated complexity exceeds a threshold.

16. Computer-readable media embodying a program of instructions executable by a computer to perform a method of managing resources of a system which performs a plurality of signal processing functions including echo cancellation function on a signal, the method comprising:

estimating average complexity of each of the signal processing functions by comparing a first signal to a second signal;

summing the estimated average complexity of the each of the signal processing functions; and

adjusting adaptation speed of the echo cancellation function by changing the number of coefficients of the echo canceller, when the sum of the estimated average complexities exceeds a threshold, wherein said estimating signal processing complexity comprises estimating maximum power level of a reference signal, long term average power of an error signal, and long term average power of a near end signal.

17. The computer-readable media of claim 16 further comprising instructions for bypassing the echo cancellation function and suppressing echo of the signal by an echo suppressor instead, when the sum of the estimated average complexities exceeds a threshold.

18. The computer-readable media of claim 16 wherein the adjusting adaptation speed of the echo cancellation function comprises reducing the complexity of the echo cancellation adaption.

Description

This application contains subject matter that is related to co-pending patent application Ser. No. 09/639,527, filed Aug. 16, 2000; co-pending patent Application No. 09/493,458, filed Jan. 28, 2000; co-pending patent application Ser. No. 09/643,920, filed Aug. 23, 2000; co-pending patent application Ser. No. 09/692,554, filed Oct. 19, 2000; co-pending patent application Ser. No. 09/644,586, filed Aug. 23, 2000; co-pending patent application Ser. No. 09/653,261, filed Aug. 31, 2000; co-pending patent application Ser. No. 09/654,376, filed Sep. 1, 2000; co-pending patent application Ser. No. 09/533,022, filed Mar. 22, 2000; co-pending patent application Ser. No. 09/697,777, filed Oct. 26, 2000; co-pending patent application Ser. No. 09/651,006, filed Aug. 29, 2000; and co-pending patent application Ser. No. 09/522,184, filed Mar. 9, 2000.

FIELD OF THE INVENTION

The present invention relates generally to telecommunications systems, and more particularly, to a system for interfacing telephony devices with packet based networks.

BACKGROUND

Telephony devices, such as telephones, analog fax machines, and data modems, have traditionally utilized circuit switched networks to communicate. With the current state of technology, it is desirable for telephony devices to communicate over the Internet, or other packet based networks. Heretofore, an integrated system for interfacing various telephony devices over packet based networks has been difficult due to the different modulation schemes of the telephony devices. Accordingly, it would be advantageous to have an efficient and robust integrated system for the exchange of voice, fax data and modem data between telephony devices and packet based networks.

SUMMARY OF THE INVENTION

In one aspect of the present invention, a method of managing resources of a system includes processing data, estimating data processing complexity, and reducing the data processing complexity when the estimated complexity exceeds a threshold.

In another aspect of the present invention, a method of managing resources of a system includes performing a plurality of system functions on data, estimating average complexity of each the system functions, summing the estimated average complexity of each of the system functions, and reducing complexity of at least one of the system functions when the sum of the estimated average complexities exceeds a threshold.

In yet another aspect of the present invention, a data transmission includes a telephony device which outputs a signal, and a signal processor coupled to the telephony device, the signal processor comprising a resource manager that estimates signal processor complexity and reduces the signal processor complexity when the estimated complexity exceeds a threshold.

In still yet another aspect of the present invention, a resource manager for a signal processor includes estimation means for estimating signal processor complexity, and reduction means for reducing the signal processor complexity when the estimated complexity exceeds a threshold.

In still yet a further aspect of the present invention, a resource manager for a signal processor performing a plurality of functions includes estimation means for estimating average complexity of each the system functions, summing means for summing the estimated average complexity of each of the system functions, and reduction means for reducing complexity of at least one of the system functions when the sum of the estimated average complexities exceeds a threshold.

In a further aspect of the present invention, computer-readable media embodying a program of instructions executable by a computer performs a method of managing resources of a system which processes data, the method including estimating data processing complexity, and reducing the data processing complexity when the estimated complexity exceeds a threshold.

In still a further aspect of the present invention, computer-readable media embodying a program of instructions executable by a computer performs a method of managing resources of a system which performs a plurality of functions on data, the method including estimating average complexity of each the system functions, summing the estimated average complexity of each of the system functions, and reducing complexity of at least one of the system functions when the sum of the estimated average complexities exceeds a threshold.

It is understood that other embodiments of the present invention will become readily apparent to those skilled in the art from the following detailed description, wherein it is shown and described only embodiments of the invention by way of illustration of the best modes contemplated for carrying out the invention. As will be realized, the invention is capable of other and different embodiments and its several details are capable of modification in various other respects, all without departing from the spirit and scope of the present invention. Accordingly, the drawings and detailed description are to be regarded as illustrative in nature and not as restrictive.

DESCRIPTION OF THE DRAWINGS

These and other features, aspects, and advantages of the present invention will become better understood with regard to the following description, appended claims, and accompanying drawings where:

FIG. 1 is a block diagram of packet based infrastructure providing a communication medium with a number of telephony devices in accordance with a preferred embodiment of the present invention;

FIG. 2 is a block diagram of a signal processing system implemented with a programmable digital signal processor (DSP) software architecture in accordance with a preferred embodiment of the present invention;

FIG. 3 is a block diagram of the software architecture operating on the DSP platform of FIG. 2 in accordance with a preferred embodiment of the present invention;

FIG. 4 is state machine diagram of the operational modes of a virtual device driver for packet based network applications in accordance with a preferred embodiment of the present invention;

FIG. 5 is a block diagram of several signal processing systems in the voice mode for interfacing between a switched circuit network and a packet based network in accordance with a preferred embodiment of the present invention;

FIG. 6 is a system block diagram of a signal processing system operating in a voice mode in accordance with a preferred embodiment of the present invention;

FIG. 7 is a block diagram of a method for canceling echo returns in accordance with a preferred embodiment of the present invention;

FIG. 8A is a block diagram of a method for normalizing the power level of a digital voice samples to ensure that the conversation is of an acceptable loudness in accordance with a preferred embodiment of the present invention;

FIG. 8B is a graphical depiction of a representative output of a peak tracker as a function of a typical input signal, demonstrating that the reference value that the peak tracker forwards to a gain calculator to adjust the power level of digital voice samples should preferably rise quickly if the signal amplitude increases, but decrement slowly if the signal amplitude decreases in accordance with a preferred embodiment of the present invention;

FIG. 9 is a graphical depiction of exemplary operating thresholds for adjusting the gain factor applied to digital voice samples to ensure that the conversation is of an acceptable loudness in accordance with a preferred embodiment of the present invention;

FIG. 10 is a block diagram of a method for estimating the spectral shape of the background noise of a voice transmission in accordance with a preferred embodiment of the present invention;

FIG. 11 is a block diagram of a method for generating comfort noise with an energy level and spectral shape that substantially matches the background noise of a voice transmission in accordance with a preferred embodiment of the present invention;

FIG. 12 is a block diagram of the voice decoder and the lost packet recovery engine in accordance with a preferred embodiment of the present invention;

FIG. 13A is a flow chart of the preferred lost frame recovery algorithm in accordance with a preferred embodiment of the present invention;

FIG. 13B is a flow chart of the voicing decision and pitch period calculation in accordance with a preferred embodiment of the present invention;

FIG. 13C is a flow chart demonstrating voicing synthesis performed when packets are lost and for the first decoded voice packet after a series of lost packets in accordance with a preferred embodiment of the present invention;

FIG. 14 is a block diagram of a method for detecting dual tone multi frequency tones in accordance with a preferred embodiment of the present invention;

FIG. 14A is a block diagram of a method for reducing the instructions required to detect a valid dual tone and for pre-detecting a dual tone;

FIG. 15 is a block diagram of a signaling service for detecting precise tones in accordance with a preferred embodiment of the present invention;

FIG. 16 is a block diagram of a method for detecting the frequency of a precise tone in accordance with a preferred embodiment of the present invention;

FIG. 17 is state machine diagram of a power state machine which monitors the estimated power level within each of the precise tone frequency bands in accordance with a preferred embodiment of the present invention;

FIG. 18 is state machine diagram of a cadence state machine for monitoring the cadence (on/off times) of a precise tone in a voice signal in accordance with a preferred embodiment of the present invention;

FIG. 18A is a block diagram of a cadence processor for detecting precise tones in accordance with a preferred embodiment of the present invention;

FIG. 19 is a block diagram of resource manager interface with several VHD's and PXD's in accordance with a preferred embodiment of the present invention;

FIG. 20 is a block diagram of several signal processing systems in the fax relay mode for interfacing between a switched circuit network and a packet based network in accordance with a preferred embodiment of the present invention;

FIG. 21 is a system block diagram of a signal processing system operating in a real time fax relay mode in accordance with a preferred embodiment of the present invention;

FIG. 22 is a diagram of the message flow for a fax relay in non error control mode in accordance with a preferred embodiment of the present invention;

FIG. 23 is a flow diagram of a method for fax mode spoofing in accordance with a preferred embodiment of the present invention;

FIG. 24 is a block diagram of several signal processing systems in the modem relay mode for interfacing between a switched circuit network and a packet based network in accordance with a preferred embodiment of the present invention;

FIG. 25 is a system block diagram of a signal processing system operating in a modem relay mode in accordance with a preferred embodiment of the present invention;

FIG. 26 is a diagram of a relay sequence for V.32bis rate synchronization using rate re-negotiation in accordance with a preferred embodiment of the present invention;

FIG. 27 is a diagram of an alternate relay sequence for V.32bis rate synchronization whereby rate signals are used to align the connection rates at the two ends of the network without rate re-negotiation in accordance with a preferred embodiment of the present invention;

FIG. 28 is a system block diagram of a QAM data pump transmitter in accordance with a preferred embodiment of the present invention;

FIG. 29 is a system block diagram of a QAM data pump receiver in accordance with a preferred embodiment of the present invention;

FIG. 30 is a block diagram of a method for sampling a signal of symbols received in a data pump receiver in synchronism with the transmitter clock of a data pump transmitter in accordance with a preferred embodiment of the present invention;

FIG. 31 is a block diagram of a second order loop filter for reducing symbol clock jitter in the timing recovery system of data pump receiver in accordance with a preferred embodiment of the present invention;

FIG. 32 is a block diagram of an alternate method for sampling a signal of symbols received in a data pump receiver in synchronism with the transmitter clock of a data pump transmitter in accordance with a preferred embodiment of the present invention;

FIG. 33 is a block diagram of an alternate method for sampling a signal of symbols received in a data pump receiver in synchronism with the transmitter clock of a data pump transmitter wherein a timing frequency offset compensator provides a fixed dc component to compensate for clock frequency offset present in the received signal in accordance with a preferred embodiment of the present invention;

FIG. 34 is a block diagram of a method for estimating the timing frequency offset required to sample a signal of symbols received in a data pump receiver in synchronism with the transmitter clock of a data pump transmitter in accordance with a preferred embodiment of the present invention;

FIG. 35 is a block diagram of a method for adjusting the gain of a data pump receiver (fax or modem) to compensate for variations in transmission channel conditions; and

FIG. 36 is a block diagram of a method for detecting human speech in a telephony signal.

DETAILED DESCRIPTION

An Embodiment of a Signal Processing System

In a preferred embodiment of the present invention, a signal processing system is employed to interface telephony devices with packet based networks. Telephony devices include, by way of example, analog and digital phones, ethernet phones, Internet Protocol phones, fax machines, data modems, cable modems, interactive voice response systems, PBXs, key systems, and any other conventional telephony devices known in the art. The described preferred 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 information, including voice, fax and modem data over packet based networks. The embedded communications software is preferably run on programmable digital signal processors (DSPs) and is used in gateways, cable modems, remote access servers, PBXs, and other packet based network appliances.

An exemplary topology is shown in FIG. 1 with a packet based network 10 providing a communication medium between various telephony devices. Each network gateway 12a, 12b, 12c includes a signal processing system which provides an interface between the packet based network 10 and a number of telephony devices. In the described exemplary embodiment, each network gateway 12a, 12b, 12c supports a fax machine 14a, 14b, 14c, a telephone 13a, 13b, 13c, and a modem 15a, 15b, 15c. As will be appreciated by those skilled in the art, each network gateway 12a, 12b, 12c could support a variety of different telephony arrangements. By way of example, each network gateway might support any number telephony devices and/or circuit switched/packet based networks including, among others, analog telephones, ethernet phones, fax machines, data modems, PSTN lines (Public Switching 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 12a, 12b provide a direct interface between their respective telephony devices and the packet based network 10. The other network gateway 12c is connected to its respective telephony device through a PSTN 19. The network gateways 12a, 12b, 12c 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.

The exemplary signal processing system can be implemented with a programmable DSP software architecture as shown in FIG. 2. This architecture has a DSP 17 with memory 18 at the core, a number of network channel interfaces 19 and telephony interfaces 20, and a host 21 that may reside in the DSP itself or on a separate microcontroller. The network channef interfaces 19 provide multi-channel access to the packet based network. The telephony interfaces 23 can be connected to a circuit switched network interface such as a PSTN system, or directly to any telephony device. The programmable DSP is effectively hidden within the embedded communications software layer. The software layer binds all core DSP algorithms together, interfaces the DSP hardware to the host, and provides low level services such as the allocation of resources to allow higher level software programs to run.

An exemplary multi-layer software architecture operating on a DSP platform is shown in FIG. 3. A user application layer 26 provides overall executive control and system management, and directly interfaces a DSP server 25 to the host 21 (see to FIG. 2). The DSP server 25 provides DSP resource management and telecommunications signal processing. Operating below the DSP server layer are a number of physical devices (PXD) 30a, 30b, 30c. Each PXD provides an interface between the DSP server 25 and an external telephony device (not shown) via a hardware abstraction layer (HAL) 34.

The DSP server 25 includes a resource manager 24 which receives commands from, forwards events to, and exchanges data with the user application layer 26. The user application layer 26 can either be resident on the DSP 17 or alternatively on the host 21 (see FIG. 2), such as a microcontroller. An application programming interface 27 (API) provides a software interface between the user application layer 26 and the resource manager 24. The resource manager 24 manages the internal/external program and data memory of the DSP 17. In addition the resource manager dynamically allocates DSP resources, performs command routing as well as other general purpose functions.

The DSP server 25 also includes virtual device drivers (VHDS) 22a, 22b, 22c. The VHDs are a collection of software objects that control the operation of and provide the facility for real time signal processing. Each VHD 22a, 22b, 22c includes an inbound and outbound media queue (not shown) and a library of signal processing services specific to that VHD 22a, 22b, 22c. In the described exemplary embodiment, each VHD 22a, 22b, 22c is a complete self-contained software module for processing a single channel with a number of different telephony devices. Multiple channel capability can be achieved by adding VHDs to the DSP server 25. The resource manager 24 dynamically controls the creation and deletion of VHDs and services.

A switchboard 32 in the DSP server 25 dynamically inter-connects the PXDs 30a, 30b, 30c with the VHDs 22a, 22b, 22c. Each PXD 30a, 30b, 30c is a collection of software objects which provide signal conditioning for one external telephony device. For example, a PXD may provide volume and gain control for signals from a telephony device prior to communication with the switchboard 32. Multiple telephony functionalities can be supported on a single channel by connecting multiple PXDs, one for each telephony device, to a single VHD via the switchboard 32. Connections within the switchboard 32 are managed by the user application layer 26 via a set of API commands to the resource manager 24. The number of PXDs and VHDs is expandable, and limited only by the memory size and the MIPS (millions instructions per second) of the underlying hardware.

A hardware abstraction layer (HAL) 34 interfaces directly with the underlying DSP 17 hardware (see FIG. 2) and exchanges telephony signals between the external telephony devices and the PXDs. The HAL 34 includes basic hardware interface routines, including DSP initialization, target hardware control, codec sampling, and hardware control interface routines. The DSP initialization routine is invoked by the user application layer 26 to initiate the initialization of the signal processing system. The DSP initialization sets up the internal registers of the signal processing system for memory organization, interrupt handling, timer initialization, and DSP configuration. Target hardware initialization involves the initialization of all hardware devices and circuits external to the signal processing system. The HAL 34 is a physical firmware layer that isolates the communications software from the underlying hardware. This methodology allows the communications software to be ported to various hardware platforms by porting only the affected portions of the HAL 34 to the target hardware.

The exemplary software architecture described above can be integrated into numerous telecommunications products. In an exemplary embodiment, the software architecture is designed to support telephony signals between telephony devices (and/or circuit switched networks) and packet based networks. A network VHD (NetVHD) is used to provide a single channel of operation and provide the signal processing services for transparently managing voice, fax, and modem data across a variety of packet based networks. More particularly, the NetVHD encodes and packetizes DTMF, voice, fax, and modem data received from various telephony devices and/or circuit switched networks and transmits the packets to the user application layer. In addition, the NetVHD disassembles DTMF, voice, fax, and modem data from the user application layer, decodes the packets into signals, and transmits the signals to the circuit switched network or device.

An exemplary embodiment of the NetVHD operating in the described software architecture is shown in FIG. 4. The NetVHD includes four operational modes, namely voice mode 36, voiceband data mode 37, fax relay mode 40, and data relay mode 42. In each operational mode, the resource manager invokes various services. For example, in the voice mode 36, the resource manager invokes call discrimination 44, packet voice exchange 48, and packet tone exchange 50. The packet voice exchange 48 may employ numerous voice compression algorithms, including, among others, Linear 128 kbps, G.711 u-law/A-law 64 kbps (ITU Recommendation G.711 (1988)—Pulse code modulation (PCM) of voice frequencies), G.726 16/24/32/40 kbps (ITU Recommendation G.726 (12/90)-40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM)), G.729A 8 kbps (Annex A (11/96) to ITU Recommendation G.729—Coding of speech at 8 kbit/s using conjugate structure algebraic-code-excited linear-prediction (CS-ACELP)—Annex A: Reduced complexity 8 kbit/s CS-ACELP speech codec), and G.723 5.3/6.3 kbps (ITU Recommendation G.723.1 (03/96)—Dual rate coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s). The contents of each of the foregoing ITU Recommendations being incorporated herein by reference as if set forth in full.

The packet voice exchange 48 is common to both the voice mode 36 and the voiceband data mode 37. In the voiceband data mode 37, the resource manager invokes the packet voice exchange 48 for exchanging transparently data without modification (other than packetization) between the telephony device (or circuit switched network) and the packet based network. This is typically used for the exchange of fax and modem data when bandwidth concerns are minimal as an alternative to demodulation and remodulation. During the voiceband data mode 37, the human speech detector service 59 is also invoked by the resource manager. The human speech detector 59 monitors the signal from the near end telephony device for speech. In the event that speech is detected by the human speech detector 59, an event is forwarded to the resource manager which, in turn, causes the resource manager to terminate the human speech detector service 59 and invoke the appropriate services for the voice mode 36 (i.e., the call discriminator, the packet tone exchange, and the packet voice exchange).

In the fax relay mode 40, the resource manager invokes a fax exchange 52 service. The packet fax exchange 52 may employ various data pumps including, among others, V.17 which can operate up to 14,400 bits per second, V.29 which uses a 1700-Hz carrer that is varied in both phase and amplitude, resulting in 16 combinations of 8 phases and 4 amplitudes which can operate up to 9600 bits per second, and V.27ter which can operate up to 4800 bits per second. Likewise, the resource manager invokes a packet data exchange 54 service in the data relay mode 42. The packet data exchange 52 may employ various data pumps including, among others, V.22bis/V.22 with data rates up to 2400 bits per second, V.32bis/V.32 which enables full-duplex transmission at 14,400 bits per second, and V.34 which operates up to 33,600 bits per second. The ITU Recommendations setting forth the standards for the foregoing data pumps are incorporated herein by reference as if set forth in full.

In the described exemplary embodiment, the user application layer does not need to manage any service directly. The user application layer manages the session using high-level commands directed to the NetVHD, which in turn directly runs the services. However, the user application layer can access more detailed parameters of any service if necessary to change, by way of example, default functions for any particular application.

In operation, the user application layer opens the NetVHD and connects it to the appropriate PXD. The user application then may configure various operational parameters of the NetVHD, including, among others, default voice compression (Linear, G.711, G.726, G.723.1, G.723.1A, G.729A, G.729B), fax data pump (Binary, V.17, V.29, V.27ter), and modem data pump (Binary, V.22bis, V.32bis, V.34). The user application layer then loads an appropriate signaling service (not shown) into the NetVHD, configures it and sets the NetVHD to the On-hook state.

In response to events from the signaling service (not shown) via a near end telephony device (hookswitch), or signal packets from the far end, the user application will set the NetVHD to the appropriate off-hook state, typically voice mode. In an exemplary embodiment, if the signaling service event is triggered by the near end telephony device, the packet tone exchange will generate dial tone. Once a DTMF tone is detected, the dial tone is terminated. The DTMF tones are packetized and forwarded to the user application layer for transmission on the packet based network. The packet tone exchange could also play ringing tone back to the near end telephony device (when a far end telephony device is being rung), and a busy tone if the far end telephony device is unavailable. Other tones may also be supported to indicate all circuits are busy, or an invalid sequence of DTMF digits were entered on the near end telephony device.

Once a connection is made between the near end and far end telephony devices, the call discriminator is responsible for differentiating between a voice and machine call by detecting the presence of a 2100 Hz. tone (as in the case when the telephony device is a fax or a modem), a 1100 Hz. tone or V.21 modulated high level data link control (HDLC) flags (as in the case when the telephony device is a fax). If a 1100 Hz. tone, or V.21 modulated HDLC flags are detected, a calling fax machine is recognized. The NetVHD then terminates the voice mode 36 and invokes the packet fax exchange to process the call. If however, 2100 Hz tone is detected, the NetVHD terminates voice mode and invokes the packet data exchange.

The packet data exchange service further differentiates between a fax and modem by continuing to monitor the incoming signal for V.21 modulated HDLC flags, which if present, indicate that a fax connection is in progress. If HDLC flags are detected, the NetVHD terminates packet data exchange service and initiates packet fax exchange service. Otherwise, the packet data exchange service remains operative. In the absence of an 1100 or 2100 Hz. tone, or V.21 modulated HDLC flags the voice mode remains operative.

A. The Voice Mode

Voice mode provides signal processing of voice signals. As shown in the exemplary embodiment depicted in FIG. 5, voice mode enables the transmission of voice 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. The voice mode should also permit voice to be carried over traditional media such as time division multiplex (TDM) networks and voice storage and playback systems. Network gateway 55a supports the exchange of voice between a traditional circuit switched 58 and a packet based network 56. In addition, network gateways 55b, 55c, 55d, 55e support the exchange of voice between the packet based network 56 and a number of telephones 57a, 57b, 57c, 57d, 57e. Although the described exemplary embodiment is shown for telephone communications across the packet based network, it will be appreciated by those skilled in the art that other telephony/network devices could be used in place of one or more of the telephones, such as a HPNA phone connected via a cable modem.

The PXDs for the voice mode provide echo cancellation, gain, and automatic gain control. The network VHD invokes numerous services in the voice mode including call discrimination, packet voice exchange, and packet tone exchange. These network VHD services operate together to provide: (1) an encoder system with DTMF detection, call progress tone detection, voice activity detection, voice compression, and comfort noise estimation, and (2) a decoder system with delay compensation, voice decoding, DTMF generation, comfort noise generation and lost frame recovery.

The services invoked by the network VHD in the voice mode and the associated PXD is shown schematically in FIG. 6. 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. DS0) carrying a 64 kb/s pulse code modulated (PCM) signal, i.e., digital voice samples.

The incoming PCM signal 60a is initially processed by the PXD 60 to remove far end echos that might otherwise be transmitted back to the far end user. As the name implies, echos in telephone systems is the return of the talker'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 loud they are talking, and indeed, without side-tone, users tend to talk too loud. 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 echos from far end speech present on the incoming PCM signal 60a before routing the incoming PCM signal 60a back to the far end user. The echo canceller 70 samples an outgoing PCM signal 60b from the far end user, filters it, and combines it with the incoming PCM signal 60a. 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 60b. 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, a 1100 Hz. tone or V.21 modulated HDLC flags are present. As described above with reference to FIG. 4, 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, 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 be disabled when the NLP 72 is activated. This is accomplished by a "NLP on" message 72a 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 outputted 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 80a 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 by reference as through 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 76a 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 76a 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 76a 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 depacketizing 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 depacketizing 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 depacketizing 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 at the same time or in the same order 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 192 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 must continue 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 98a 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 depacketized by the depacketizing engine 84. DTMF frames at the output of the depacketizing 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 88a 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 104a 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 depacketized by the depacketizing engine 84. Call progress tone frames at the output of the depacketizing 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 87a 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 60b.

1. Echo Canceller with NLP

The problem of line echos such as the reflection of the talker's voice resulting from the operation of the hybrid with its two-four wire conversion is a common telephony problem. To eliminate or minimize the effect of line echos in the described exemplary embodiment of the present invention, an echo canceller with non-linear processing is used. Although echo cancellation is described in the context of a signal processing system for packet voice exchange, those skilled in the art will appreciate that the techniques described for echo cancellation are likewise suitable for various applications requiring the cancellation of reflections, or other undesirable signals, from a transmission line. Accordingly, the described exemplary embodiment for echo cancellation in a signal processing system is by way of example only and not by way of limitation.

In the described exemplary embodiment the echo canceller preferably complies with one or more of the following ITU-T Recommendations G.164 (1988)—Echo Suppressors, G.165 (March 1993)—Echo Cancellers, and G.168 (April 1997)—Digital Network Echo Cancellers, the contents of which are incorporated herein by reference as though set forth in full. The described embodiment merges echo cancellation and echo suppression methodologies to remove the line echos that are prevalent in telecommunication systems. Typically, echo cancellers are favored over echo suppressors for superior overall performance in the presence of system noise such as, for example, background music, double talk etc., while echo suppressors tend to perform well over a wide range of operating conditions where clutter such as system noise is not present. The described exemplary embodiment utilizes an echo suppressor when the energy level of the line echo is below the audible threshold, otherwise an echo canceller is preferably used. The use of an echo suppressor reduces system complexity, leading to lower overall power consumption or higher densities (more VHDs per part or network gateway). Those skilled in the art will appreciate that various signal characteristics such as energy, average magnitude, echo characteristics, as well as information explicitly received in voice or SID packets may be used to determine when to bypass echo cancellation. Accordingly, the described exemplary embodiment for bypassing echo cancellation in a signal processing system as a function of estimated echo power is by way of example only and not by way of limitation.

FIG. 7 shows the block diagram of an echo canceller in accordance with a preferred embodiment of the present invention. If required to support voice transmission via a T1 or other similar transmission media, a compressor 120 may compress the output 120(a) of the voice decoder system into a format suitable for the channel at R_out 120(b). Typically the compressor 120 provides μ-law or A-law compression in accordance with ITU-T standard G.711, although linear compression or compression in accordance with alternate companding laws may also be supported. The compressed signal at R_out (signal that eventually makes it way to a near end ear piece/telephone receiver), may be reflected back as an input signal to the voice encoder system. An input signal 122(a) may also be in the compressed domain (if compressed by compressor 120) and, if so, an expander 122 may be required to invert the companding law to obtain a near end signal 122(b). A power estimator 124 estimates a short term average power 124(a), a long term average power 124(b), and a maximum power level 124(c) for the near end signal 122(b).

An expander 126 inverts the companding law used to compress the voice decoder output signal 120(b) to obtain a reference signal 126(a). One of skill in the art will appreciated that the voice decoder output signal could alternatively be compressed downstream of the echo canceller so that the expander 126 would not be required. However, to ensure that all non-linearities in the echo path are accounted for in the reference signal 126(a) it is preferable to compress/expand the voice decoder output signal 120(b). A power estimator 128 estimates a short term average power 128(a), a long term average power 128(b), a maximum power level 128(c) and a background power level 128(d) for the reference signal 126(a). The reference signal 126(a) is input into a finite impulse response (FIR) filter 130. The FIR filter 130 models the transfer characteristics of a dialed telephone line circuit so that the unwanted echo may preferably be canceled by subtracting filtered reference signal 130(a) from the near end signal 122(b) in a difference operator 132.

However, for a variety of reasons, such as for example, non-linearities in the hybrid and tail circuit, estimation errors, noise in the system, etc., the adaptive FIR filter 130 may not identically model the transfer characteristics of the telephone line circuit so that the echo canceller may be unable to cancel all of the resulting echo. Therefore, a non linear processor (NLP) 140 is used to suppress the residual echo during periods of far end active speech with no near end speech. During periods of inactive speech, a power estimator 138 estimates the performance of the echo canceller by estimating a short term average power 138(a), a long term average power 138(b) and background power level 138(c) for an error signal 132(b) which is an output of the difference operator 132. The estimated performance of the echo canceller is one measure utilized by adaptation logic 136 to selectively enable a filter adapter 134 which controls the convergence of the adaptive FIR filter 130. The adaptation logic 136 processes the estimated power levels of the reference signal (128a, 128b, 128c and 128d) the near end signal (124a, 124b and 124c) and the error signal (138a, 138b and 138c) to control the invocation of the filter adapter 134 as well as the step size to be used during adaptation.

In the described preferred embodiment, the echo suppressor is a simple bypass 144(a) that is selectively enabled by toggling the bypass cancellation switch 144. A bypass estimator 142 toggles the bypass cancellation switch 144 based upon the maximum power level 128(c) of the reference signal 126(a), the long term average power 138(b) of the error signal 132(b) and the long term average power 124(b) of the near end signal 122(b). One skilled in the art will appreciate that a NLP or other suppressor could be included in the bypass path 144(a), so that the described echo suppressor is by way of example only and not by way of limitation.

In an exemplary embodiment, the adaptive filter 130 models the transfer characteristics of the hybrid and the tail circuit of the telephone circuit. The tail length supported should preferably be at least 16 msec. The adaptive filter 130 may be a linear transversal filter or other suitable finite impulse response filter. In the described exemplary embodiment, the echo canceller preferably converges or adapts only in the absence of near end speech. Therefore, near end speech and/or noise present on the input signal 122(a) may cause the filter adapter 134 to diverge. To avoid divergence the filter adapter 134 is preferably selectively enabled by the adaptation logic 136. In addition, the time required for an adaptive filter to converge increases significantly with the number of coefficients to be determined. Reasonable modeling of the hybrid and tail circuits with a finite impulse response filter requires a large number of coefficients so that filter adaptation is typically computationally intense. In the described exemplary embodiment the DSP resources required for filter adaptation are minimized by adjusting the adaptation speed of the FIR filter 130.

The filter adapter 134 is preferably based upon a normalized least mean square algorithm (NLMS) as described in S. Haykin, Adaptive Filter Theory, and T. Parsons, Voice and Speech Processing, the contents of which are incorporated herein by reference as if set forth in full. The error signal 132(b) at the output of the difference operator 132 for the adaptation logic may preferably be characterized as follows:
##EQU1##