Voice-identification-based signal processing for multiple-talker applications

The audio signals associated with different co-located groups of talkers in a teleconference are detected (e.g., by comparing the voiceprint for the current talker group with stored voiceprints corresponding to all of the co-located teleconference participants) and processed using different and appropriate automatic gain control (AGC) levels, where each group has a corresponding stored AGC level. Depending on the embodiment, each group may have one or more participants.

BACKGROUND OF THE INVENTION

1. Field of the Invention

This invention relates generally to telecommunications systems, and more particularly to audio signal processing for teleconferencing systems where multiple talkers share a sound transceiver.

2. Description of the Related Art

Teleconferences often have multiple talkers at a single site. Such talkers typically share a sound transceiver. A typical shared sound transceiver includes a microphone and a speaker. The sound received at the microphone is converted into an electrical signal for processing and transmission to remotely located teleconferencing participants via suitable electrical, optical, and/or wireless communication networks. The elements processing the electrical signal typically include an automatic gain control (AGC) circuit, which adjusts, by appropriate amplification or attenuation, the electrical signal so that the amplitude of the adjusted electrical signal is generally within some acceptable range. This is done so that the adjusted signal's amplitude is neither too high nor too low. The adjusted signal may undergo additional processing steps to improve sound quality (e.g., reduce noise) or increase transmission efficiency, and may be added to one or more other signals for transmission to the remote locations. Typically, the adjusted signal is eventually received at sound transceivers of the other participants and converted back to sound.

The AGC processing of the electrical signal corresponding to the received sound is typically based on the composite and time-averaged characteristics of the received sound. When a teleconference having multiple participants at a single site includes, for example, a loud talker positioned close to the microphone and a soft talker positioned far from the microphone, AGC processing based on the time-averaged characteristics of both talkers will tend to insufficiently amplify signals corresponding to the soft talker and insufficiently attenuate signals corresponding to the loud talker, resulting in less than desirable playback at the remote locations.

SUMMARY OF THE INVENTION

One embodiment of the invention is a method of processing an audio signal, the method comprising: (a) processing the audio signal to identify a first reference voiceprint; (b) assigning an AGC level to the first reference voiceprint; and (c) applying the AGC level assigned to the first reference voiceprint to AGC processing of a first portion of the audio signal corresponding to the first reference voiceprint. Another embodiment of the invention is an audio processing apparatus comprising: control logic adapted to (a) process the audio signal to identify a first reference voiceprint and (b) assign an AGC level to the first reference voiceprint; and a signal processor adapted to apply the AGC level assigned to the first reference voiceprint to AGC processing of a first portion of the audio signal corresponding to the first reference voiceprint.

DETAILED DESCRIPTION

In a preferred embodiment, the present invention is a system that identifies a current talker based on the talker's voiceprint, finds or assigns an AGC level for the current talker, and applies that AGC level to AGC processing of signals corresponding to the talker. The system can store voiceprints and corresponding AGC levels during a training mode, or do so on the fly as talkers talk during a conference call.

FIG. 1shows a simplified block diagram of a communications system100, according to an exemplary embodiment of the present invention implementing distributed processing such that each local conference node processes its own outgoing signal. As shown inFIG. 1, communications system100includes two (or more) conference nodes (101,102) communicating via communication network cloud107. Each conference node comprises a microphone (103,110), a voice-processing module (104,111), a speaker (106,109), and a communication device (105,108).

Microphone103of conference node101captures the ambient sound, which may include the voice of a current talker and any background noise, and converts the sound into a corresponding electrical signal104a. Signal104ais received by voice-processing module104, which processes the signal in accordance with the present invention. Voice-processing module104outputs processed electrical signal105ato communications device105. Communication device105interfaces with communication network cloud107, to which it transmits signal107a, which includes a signal substantially corresponding to signal105a. Communication network cloud107transmits and processes signal107ain any appropriate way that allows it to output to remote conference node102a signal108athat includes a signal substantially identical to signal107a. Communication device108of conference node102receives and processes signal108aand outputs signal109ato speaker109, which converts signal109ato audible sound so that a listener near speaker109would hear the current talker in the vicinity of microphone103in accordance with this embodiment of the invention. Analogous processing is performed by corresponding elements for audio signal processing and transmission from conference node102to conference node101.

FIG. 2shows a block diagram of voice-processing module200, which may be used to implement each of voice-processing modules104and111ofFIG. 1. Voice-processing module200includes A/D converter201, voice processor203, signal processor202, control logic204, database205, and user interface206. A/D converter201receives analog signal201afrom a microphone (e.g.,103or110ofFIG. 1) and generates corresponding digital signal202a. Note that, if the microphone generates digital signals, then A/D converter201may be omitted. Digital signal202ais input into signal processor202and voice processor203. Voice processor203generates voiceprints corresponding to continuous samples of the received signal in accordance with any of the methods known in the art, such as by generating parameters from spectrum, cepstrum, and/or other analysis of the signal.

The voiceprints generated by voice processor203are transmitted via signal204ato control logic204, which, for each voiceprint, determines whether there is a matching voiceprint in database205. Database205is adapted to store voiceprints and affiliated AGC levels; it communicates with control logic204via signal205a. If control logic204finds a matching voiceprint to the current voiceprint in database205, then the associated AGC level is retrieved by control logic204, and may be stored in a cache-like memory together with the retrieved or received voiceprint, which serves as a reference voiceprint. Control logic204then transmits the retrieved AGC level via signal204bto signal processor202, which comprises an AGC circuit. Signal processor202applies the AGC level received from control logic204during its AGC processing of signal202a, and transmits the output as signal202b. The reference voiceprint can be compared to the voiceprints being generated by voice processor203so that there is no need to search the database while the generated voiceprint matches the reference voiceprint. While the current talker keeps talking, the AGC circuit may adjust its AGC level in accordance with the current talker's signal characteristics. If the current talker is done talking (for example, when control logic204determines that a new talker is talking), then control logic204may retrieve the current talker's last AGC level from signal processor202via signal204bto update the corresponding entry in database205with this AGC level.

If control logic204does not find a voiceprint in affiliated database205that matches the voiceprint of the current talker, then control logic204generates an AGC level for the current voiceprint by, for example, retrieving the current AGC level from signal processor202, and stores the voiceprint and the associated AGC level as a new entry in database205. The generated voiceprint can then be used as a reference voiceprint. Segments of the audio signal that contain only background noise will cause the generation of one or more background noise voiceprints. The affiliated AGC levels, which will be used in the processing of the audio signal corresponding to background noise, should be assigned so as to reduce the amplification of background noise. Since background noise is likely to appear frequently in the audio signal, its voiceprint can be used as a second reference voiceprint and stored in a local cache for faster comparisons. In addition to the automatic processing described, a user can adjust the AGC level associated with the current voiceprint through user interface206. User interface206may also be used to initialize and control operating and/or training modes of voice-processing module200, or to provide information about the current talker or talker group.

FIG. 3shows a flowchart for system200ofFIG. 2, in accordance with one embodiment of the present invention. First, the system is initialized (step301). Initialization301can be induced by powering on the system, resetting it, or otherwise indicating that a new operational phase is about to commence. Initialization301can occur automatically, or can be triggered manually by a user through user interface206ofFIG. 2. Next, the system determines whether training is necessary (step302). The system may determine that training is necessary if, for example, it checks for a voiceprint database and finds it to be empty, or fails to find one. The system may, for example, be set to automatically enter training mode, or to automatically skip training mode. If training is necessary, then the system enters the training module (step303). An example of a training module in accordance with an embodiment of the present invention appears inFIG. 4. After the system exits the training module, the system enters the operating module (step304). If the system determines that training is not necessary in step302, then the system enters the operating module directly from step302. An example of an operating module in accordance with an embodiment of the present invention appears inFIG. 5.

FIG. 4shows a flowchart for an optional training module303ofFIG. 3. In a preferred embodiment of the training module, each talker introduces himself or herself, thereby providing the requisite voice samples for the generation of voiceprints. First, the training module is initialized (step401), which can be accomplished as described above in relation to step301ofFIG. 3. Next, the system samples a first talker's voice (step402). Next, voice processor203ofFIG. 2analyzes the voice sample to generate a characteristic voiceprint, which control logic204stores in database205(step403). The voiceprint contains information characterizing the talker's voice and timbre. This information is used to distinguish among talkers and identify each talker when he or she talks. Next, control logic204determines and assigns an AGC level for the voiceprint, e.g., based on the corresponding processing of the AGC circuit in signal processor202, and stores that AGC level in database205so that it is associated with its respective voiceprint (step404).

Next, control logic204determines whether it needs to train for another voice (step405). Control logic204can make this determination automatically by continuously sampling, analyzing, and processing the input signal until some terminating event occurs. A terminating event can occur, for example, if a user, using user interface206, manually indicates that training mode is over, or if control logic204determines that a talker is talking continuously for a time longer than some set limit, or if control logic204determines that a new talker is in fact a prior talker (e.g., if anyone but the last talker starts talking after everyone has introduced him or herself). Control logic204can also, for example, determine whether to train for another voice based on user input through user interface206. If control logic204determines it needs to train for another voice, then steps402-405are repeated again. Thus, control logic204loops through steps402-405for as long as training is necessary. If control logic204determines it does not need to train for another voice, then the training mode ends and control logic204exits the training module (step406).

FIG. 5shows a flowchart for operating module304ofFIG. 3. The operating module starts at step501. This is typically accomplished by the termination of training module303ofFIG. 3, or by the determination that training is not necessary (step302ofFIG. 3). Next, the current talker's voice is sampled (step502). Voice processor203ofFIG. 2analyzes the sample to create a voiceprint (step503). Control logic204searches for a matching voiceprint in database205(step504). Periods when no one is talking, i.e., samples containing only background noise, are given their own voiceprints. These background noise voiceprints can be distinguished from talker voiceprints since their spectral characteristics are distinguishable from speech spectral characteristics through means well known in the art. AGC levels for background noise voiceprints are assigned differently than for talker voiceprints since it is generally desirable to attenuate background sounds rather than amplify them.

If control logic204does not find a matching voiceprint (step505), then control logic204determines that a new talker is talking, and control logic204determines an appropriate AGC level and stores the voiceprint and the corresponding AGC level in the database (step506). Processing then proceeds to step507. In an alternative embodiment (not illustrated), if control logic204does not find a matching voiceprint in step505, then control logic204does not instruct signal processor202to change the current AGC level in step507.

If a matching voiceprint is found in step505, then processing proceeds directly to step507, in which control logic204sets the current AGC level of signal processor202to the AGC level associated with the voiceprint of the current talker, and saves the corresponding voiceprint as a reference voiceprint in a local cache. In one possible embodiment (not illustrated), control logic204proceeds from step507directly to step502without saving a reference voiceprint, and continues from there. However, since that embodiment would require frequent searching of the database (step504), in a preferred embodiment, the system continues to sample the audio signal (step508), analyze it to create a voiceprint (step509), and then determine whether the voice is new, i.e., whether a new talker is speaking (step510), e.g., by having control logic204compare the current voiceprint with the reference voiceprint stored in its local cache. The system continues to repeat steps508-510until control logic204determines that a new talker is speaking in step510. Until then, while the system repeats steps508-510, signal processor202continues to perform AGC processing without receiving any new AGC level from control logic204. Signal processor202will typically adjust the AGC level over time as part of its normal AGC processing.

If control logic204determines that a new speaker is talking in step510(e.g., the current voiceprint is sufficiently different from the reference voiceprint), then the process returns to step504to search for a matching voiceprint in database205. A single person can cause sufficiently distinct voiceprints to be generated at different times, and therefore be identified as more than one talker if, for example, that person changes location relative to the microphone, or sufficiently changes his or her speaking tone at the different times.

The operations ofFIGS. 3-5enable system101ofFIG. 1to process audio signals corresponding to different talkers using different, appropriate AGC levels. For example, the signals corresponding to a loud talker positioned close to a microphone will be processed using a different AGC level from that used to process the signals corresponding to a soft talker positioned far from that microphone. In that way, the loud talker's signals will be appropriately attenuated, while the soft talker's signals will be appropriately amplified. The result will be an improved audio playback at the remote conference nodes in which both loud and soft talkers will be better able to be heard and understood.

FIG. 6shows a simplified block diagram of communication system600according to an alternative embodiment of the present invention with centralized processing such that the signal processing in accordance with the invention is performed by a centralized, shared voice-processing module607affiliated with communication network cloud606. Teleconference nodes601and602comprise elements analogous to the elements, inFIG. 1, of teleconference nodes101and102, respectively, but without a local voice-processing module, however, with means, as are known in the art, for interfacing with any user interface element of voice-processing module607(e.g., user interface206inFIG. 2).

For example, microphone603of node601converts an audio signal corresponding to a talker into electrical signal604a, which goes into communication device604. Communication device604processes signal604a, performing functions such as amplification, equalization, noise reduction, etc., and then transmits signal606a, which includes a signal corresponding to signal604a, to communication network cloud606. Network cloud606provides signal607a, a signal corresponding to signal606a, to voice-processing module607, which operates substantially similarly to voice-processing module200ofFIG. 2, to process signal606a. Voice-processing module607then outputs signal607bto communications network cloud606, which in turn provides corresponding signal608ato communication device608of teleconference node602, wherein speaker609outputs an audio signal enhanced in accordance with an embodiment of the present invention. A similar path operates in reverse, starting with microphone610in node602and going to speaker605in node601.

The above descriptions are of a preferred embodiment; however, many variations are possible which do not depart from the present invention. For example, in an alternative embodiment, if control logic204ofFIG. 2does not find a matching voiceprint, then it determines a new AGC setting based on characteristics (e.g., amplitude) of the current talker's audio signal or, alternatively, it sets the signal processor's AGC level to a default AGC level. The structures can be replicated and combined as necessary to account for multiple sites or multiple microphones at a site. Thus, for example, additional microphones can be connected to voice-processing module104, and processed signals from other systems can be combined with signal107ain communications network cloud107ofFIG. 1, or with signal606ain communication network cloud606ofFIG. 6. Not all nodes in a distributed embodiment of the invention need to use the invention; a teleconference can connect any number of nodes that include a voice-processing module embodying the invention and any number of nodes that do not. Communications signals such as107a,108a,606a, and608acan be electrical, optical, wireless, or other format, and can be analog or digital. Additional variations as known to one of ordinary skill in the art are possible and do not depart from the claims of the present invention.

In another alternative embodiment, microphone103ofFIG. 1(and other microphones) can output digital signals thus making A/D converter201ofFIG. 2unnecessary. Also, signal processor202can be analog or digital and can get its input directly from a microphone (e.g.,103or110), or from intermediary devices. Some elements described, such as the local cache in control logic204, are part of a preferred embodiment, but are not required for operation. Signal processor202can comprise adjustable signal processing circuits in addition to an AGC circuit, and settings for these circuits can be set, stored, retrieved, and used in substantially the same way as described for the AGC level. The various settings for signal processor202can be treated in aggregate as a signal processing profile that includes at least the AGC level.

In another alternative embodiment, a voiceprint identifies a group of one or more individual talkers. Individuals with sufficiently similar speech characteristics are treated as a talker group and share a voiceprint and the associated AGC level. This can be accomplished, for example, by lowering the resolution of the voiceprint generated by voice processor203, by employing a different process to generate voiceprints, such as using processing parameters generated by an audio encoder, or by altering the matching algorithm used by control logic204. Under this alternative embodiment, group-describing information could be associated with a talker type. Just as one person can be identified at different times as different talkers, so too a person could belong to more than one talker group. Thus, a talker group can contain more than one person, and a person can belong to more than one talker group.

In another alternative embodiment of training module303ofFIG. 4(not illustrated), the processing characteristics can be assigned or adjusted manually by a user. This allows a user to override the automatic settings to ensure, for example, that a particular talker or talker type is heard more loudly than others.

Embodiments of the present invention have been described using certain functional units. These units were chosen for convenience of description and do not represent the only way to organize the units' multiple sub-functions. A sub-function described as performed by one functional unit can typically be performed by other functional units. The signal paths drawn and described are not exclusive and can be altered without material modification of function. The communication signals used can generally be in any medium (e.g., electrical, electromagnetic, or optical) and in any domain (e.g., digital or analog), although the signals used among the elements of voice-processing module200are preferably electrical and digital. Furthermore, the talkers, individual or group, can be any sound-producing systems, whether human vocal chords, other living beings, electrical, mechanical, musical, or other sound generating devices.

The present invention may be implemented as circuit-based processes, including possible implementation as a single integrated circuit (such as an ASIC or an FPGA), a multi-chip module, a single card, or a multi-card circuit pack. As would be apparent to one skilled in the art, various functions of circuit elements may also be implemented as processing blocks in a software program. Such software may be employed in, for example, a digital signal processor, micro-controller, or general-purpose computer.

Although the elements in the following method claims, if any, are recited in a particular sequence with corresponding labeling, unless the claim recitations otherwise imply a particular sequence for implementing some or all of those elements, those elements are not necessarily intended to be limited to being implemented in that particular sequence. Likewise, additional steps may be included in such methods, and certain steps may be omitted or combined, in methods consistent with various embodiments of the present invention.