CN108141694A - The event detection of playback management in audio frequency apparatus - Google Patents

Abstract

According to an embodiment of the present disclosure, a method for processing audio information in an audio device may include reproducing the audio information by generating an audio output signal for delivery to at least one transducer of the audio device, receiving a signal represented in the audio device at least one input signal of external ambient sound, detecting near-field sound in the ambient sound based on the at least one input signal, and modifying characteristics of audio information reproduced to the at least one transducer in response to detecting the near-field sound .

Description

Event detection for playback management in audio devices

Related Application Cross Reference

This disclosure claims priority to U.S. Nonprovisional Patent Application Serial No. 15/229,429 filed August 5, 2016, which claims U.S. Provisional Patent Application Serial No. 15/229,429 filed August 7, 2015 Priority of Serial No. 62/202,303, U.S. Provisional Patent Application Serial No. 62/237,868, filed October 6, 2015, and U.S. Provisional Patent Application Serial No. 62/351,499, filed June 17, 2016, each Incorporated herein by reference.

technical field

The field of representative embodiments of the present disclosure relates to methods, apparatus or implementations relating to or relating to playback management in audio devices. Applications include detection of certain ambient events, but are not limited to applications relating to near-field sound detection using spatial processing based on signals received from multiple microphones, near-range sound detection, and tone alarm detection.

Background technique

Personal audio devices have become ubiquitous and used in a variety of surroundings. Headphones used in these audio devices have been so advanced that occlusion due to passive or active methods prevents the user from tracking the ambient sound field outside the audio device. Although increased isolation and uninterrupted listening are preferable in most cases, sometimes it is unavoidable for users to hear some specific surrounding events and take appropriate actions in response to them for security or enhanced user experience. For example, if a user is listening to music through his headphones and is interrupted by someone trying to start a conversation with him or her, it may be difficult to maintain the conversation unless the user pauses or reduces the volume of the playback signal. For example, US Patent No. 7,903,825 proposes an audio device in which the playback signal is modified according to the surrounding sound field. As another example, US Patent No. 8,804,974 teaches ambient event detection in personal audio devices, which can then be used to enable event-based modification of playback content. The aforementioned references also teach the use of microphones to detect various acoustic events. As another example, US Application Serial No. 14/324,286, filed July 7, 2014, teaches using a speech detector as an event detector to adjust the playback signal during a conversation. As another example, U.S. Patent No. 8,565,446 teaches the use of direction-of-arrival (DOA) estimates and interference-to-desired (near-field) speech signal ratio estimates from a set of multiple microphones to detect desired speech in the presence of non-stationary background noise to control Speech enhancement algorithm in Noise Reduction Echo Cancellation (NREC) systems. Likewise, US Application Serial No. 13/199,593 teaches that the maximum value of a normalized cross-correlation statistic obtained by cross-correlation analysis of multiple microphones can be an effective discriminator for detecting near-field speech. In US Patent No. 8,126,706 a spectral flatness measure based music detector for NREC systems is proposed to distinguish background noise from the presence of background music. US Patent No. 7,903,825, US Patent No. 8,804,974, US Application Serial No. 14/324,286, US Patent No. 8,565,446, US Application Serial No. 13/199,593, and US Patent No. 8,126,706 are incorporated herein by reference.

Contents of the invention

In accordance with the teachings of the present disclosure, one or more disadvantages and problems associated with existing methods of event detection for playback management in personal audio devices may be reduced or eliminated.

According to an embodiment of the present disclosure, a method for processing audio information in an audio device may include reproducing the audio information by generating an audio output signal for delivery to at least one transducer of the audio device, receiving a signal represented in the audio device at least one input signal of external ambient sound, detecting near-field sound from the ambient sound based on the at least one input signal, and modifying characteristics of audio information reproduced to the at least one transducer in response to detecting the near-field sound .

According to these and other embodiments of the present disclosure, an integrated circuit for implementing at least a portion of an audio device may include an audio output configured to generate an audio output signal for delivery to at least one transducer of the audio device by to reproduce audio information; a microphone input configured to receive an input signal representing ambient sound outside the audio device; a processor configured to detect near-field sound in the ambient sound according to the input signal, in response to detecting the near-field sound , to modify the characteristics of the audio information.

According to these and other embodiments of the present disclosure, a method for processing audio information in an audio device may include reproducing the audio information by generating an audio output signal for delivery to at least one transducer of the audio device, receiving a representation At least one input signal of ambient sound external to the audio device, detecting an audio event based on the at least one input signal, modifying the reproduction of the audio information to the at least one transducer in response to detecting the audio event for at least a predetermined time characteristic.

According to these and other embodiments of the present disclosure, an integrated circuit for implementing at least a portion of an audio device may include an audio output configured to generate an audio output signal for delivery to at least one transducer of the audio device by to reproduce audio information; the microphone input is configured to receive an input signal representing ambient sound outside the audio device; the processor is configured to detect an audio event based on the input signal, and modify the audio event in response to detecting the audio event for at least a predetermined time A characteristic of the audio information reproduced to the at least one transducer.

The technical advantages of the present disclosure are readily apparent to those of ordinary skill in the art from the drawings, descriptions, and claims included herein. The objects and advantages of the embodiments will be realized and attained by at least the elements, features and combinations particularly pointed out in the claims.

It is to be understood that both the foregoing general description and the following detailed description are exemplary and explanatory and are not restrictive of the scope of the claims as set forth in the present disclosure.

Description of drawings

A more complete understanding of embodiments of the invention and some of the advantages thereof may be obtained by referring to the following description taken in conjunction with the accompanying drawings, wherein like reference numerals represent like features, in which:

Figure 1 illustrates an example of a use case scenario in which such detectors can be used in conjunction with a playback management system to enhance user experience, according to an embodiment of the present disclosure;

2 illustrates an exemplary playback management system that modifies playback signals based on decisions of event detectors according to an embodiment of the present disclosure;

Figure 3 illustrates an exemplary event detector according to an embodiment of the present disclosure;

FIG. 4 illustrates functional blocks of a system for obtaining near-field spatial statistics that can be used to detect audio events according to an embodiment of the present disclosure;

FIG. 5 illustrates exemplary fusion logic for detecting near-field sound according to an embodiment of the present disclosure;

FIG. 6 illustrates exemplary fusion logic for detecting close-range sounds according to an embodiment of the present disclosure;

Figure 7 illustrates an embodiment of a proximity speech detector according to an embodiment of the present disclosure;

Figure 8 illustrates exemplary fusion logic for detecting tone alarm events according to an embodiment of the present disclosure;

9 illustrates an exemplary timing diagram illustrating delay and hysteresis logic that may be applied to a transient audio event detection signal to generate a verified audio event signal, according to an embodiment of the present disclosure;

Figure 10 illustrates different audio event detectors with delay and hysteresis logic according to an embodiment of the disclosure.

Detailed ways

According to embodiments of the present disclosure, systems and methods are presented that can use at least three different audio event detectors that can be used in an automatic playback management framework. Such audio event detectors for an audio device may include a near-field detector that detects when near-field sounds from an audio device are detected, such as when a user (e.g., is wearing or otherwise using the audio device) user of the audio device) speaks; a proximity detector that detects when a proximity sound of the audio device is detected, such as when another person near the user of the audio device speaks; a tone alarm detector , the Tone Alarm Detector detects acoustic alarms that may have occurred in the vicinity of audio equipment. Figure 1 illustrates an example of a use case scenario in which such a detector may be used in conjunction with a playback management system to enhance user experience, according to an embodiment of the present disclosure.

FIG. 2 illustrates an exemplary playback management system that modifies the playback signal based on the decision of the event detector 2 according to an embodiment of the present disclosure. Signal processing functions in the processor 50 may include an acoustic echo canceller 1 that cancels the acoustic echo at the microphone 52 due to the echo coupling between the output audio transducer 51 (e.g., a speaker) and the microphone 52. Acoustic echo received. The echo-reduced signal may be passed to event detector 2, which may detect one or more different ambient events, including but not limited to near-field events detected by near-field detector 3 (e.g., including but not limited to voice from the user of the audio device), proximity events detected by the proximity detector 4 (e.g., including but not limited to speech or other ambient sounds other than near-field sounds) and/or detected by the alarm detector 4 5 detected tone alarm events. If an audio event is detected, the event-based playback control 6 may modify the characteristics of the audio information reproduced to the output audio transducer 51 (shown as "playback content" in FIG. 2 ). Audio information may include any information reproducible at the output audio transducer 51, including, but not limited to, downlink speech associated with a telephone conversation received via a communications network (e.g., a cellular network) and/or from internal audio Internal audio of a source (eg, music file, video file, etc.).

Figure 3 illustrates an exemplary event detector according to an embodiment of the disclosure. As shown in Figure 3, an exemplary event detector may include a voice activity detector 10, a music detector 9, a direction of arrival estimator 7, a near-field spatial information extractor 8, a background noise sound pressure level estimator 11, and a decision fusion logic device 12 . The decision fusion logic 12 uses information from the voice activity detector 10, music detector 9, direction of arrival estimator 7, near-field spatial information extractor 8, and background noise sound pressure level estimator 11 to detect audio events, including But not limited to near-field sounds, proximity sounds other than near-field sounds, and tone alarms.

The near-field detector 3 can detect near-field sounds including speech. It may be desirable to modify the audio information reproduced to the output audio transducer 51 when such near-field sounds are detected, since the detection of near-field sounds may indicate that the user is engaging in a conversation. Such near-field detection may need to be able to detect near-field sound in noisy conditions, and adapt to the detection of near-field sound under very diverse background noise conditions (for example, background noise in a restaurant, noise while driving a car, etc.) error detection. As explained in more detail below, near-field detection may require the use of multiple microphones 51 for spatial sound processing. In some embodiments, such near-field sound detection may be accomplished in the same or similar manner as described in US Patent No. 8,565,446 and/or US Application Serial No. 13/199,593.

The proximity detector 4 may detect surrounding sounds (for example, voices from people close to the user, background music, etc.) other than near-field sounds. As explained in more detail below, the proximity detector can utilize the music detector and the noise sound pressure level estimate to deactivate the proximity of proximity detector 4 because it may be difficult to distinguish proximity sounds from non-stationary background noise and background music. distance detection to avoid bad user experience due to false detection of close-range sounds. In some embodiments, such proximity sound detection may be accomplished in the same or similar manner as described in US Patent No. 8,126,706, US Patent No. 8,565,446, and/or US Application Serial No. 13/199,593.

The tone alarm detector 5 may detect tone alarms (eg, sirens) approaching the audio device. In order to provide a maximum user experience, it may be desirable for the tone alarm detector 5 to ignore certain alarms (eg, alarms of low or low volume). As explained in more detail below, tone alarm detection may require spatial sound processing using multiple microphones 51 . In some embodiments, such proximity sound detection may be accomplished in the same or similar manner as described in US Patent No. 8,126,706 and/or US Application Serial No. 13/199,593.

FIG. 4 illustrates functional blocks of a system for deriving near-field spatial statistics that can be used to detect audio events according to an embodiment of the present disclosure. Sound pressure level analysis 41 may be performed on microphone 52 by estimating the inter-microphone sound pressure level difference (imd) between the near and far microphones (eg, as described in US Application Serial No. 13/199,593). Cross-correlation analysis 13 may be performed on signals received by microphone 52 to obtain direction of arrival information DOA for ambient sound impinging on microphone 52 (eg, as described in US Patent No. 8,565,446). In cross-correlation analysis 13, the maximum normalized correlation value normMaxCorr can also be obtained (eg, as described in US Application Serial No. 13/199,593). Voice activity detector 10 may detect the presence of speech and generate a signal speechDet indicative of the presence or absence of speech in ambient sound (e.g., as in the Probabilistic-Based Speech Presence/Absence-Based Method of U.S. Patent No. 7,492,889 described). Beamformer 15 may generate near-field signal estimates and interfering signal estimates based on signals from microphones 52, which may be used by noise analysis 14 to determine the noise level noiseLevel and Interference vs. near-field signal ratio idr. US Patent No. 8,565,446 describes an example method for estimating the interference-to-near-field signal ratio idr using a pair of beamformers 15 . The voice activity detector 36 may use the interference estimate to detect any voice signal (proxSpeechDet) that does not originate from the desired signal direction. The noise analysis 14 may be performed by updating the interfering signal energy based on the estimated DOA of the direction of arrival DOA of the ambient sound as long as it is outside the acceptance angle of the near-field sound. The direction of arrival of near-field sound may be known a priori for a given microphone array configuration in the industrial design of personal audio devices.

The various statistics generated by the system of FIG. 4 can then be used to detect the presence of near-field sound. FIG. 5 illustrates exemplary fusion logic for detecting near-field sound according to an embodiment of the disclosure. As shown in Figure 5, near-field speech can be detected when all of the following criteria are met:

Direction of Arrival estimation DOA of ambient sound is within the acceptance angle of near-field sound (block 16);

The maximum normalized cross-correlation statistic normMaxCorr is greater than the threshold normMaxCorrThres1 (block 17);

The interference to near-field desired signal ratio idr is less than the threshold idrThres1 (block 18);

Speech activity is detected, as indicated by the signal speechDet (block 19);

• The inter-microphone sound pressure level difference statistic imd is greater than a threshold imdTh (block 42).

In some embodiments, the thresholds idrThres and imdTh may be dynamically adjusted based on background noise sound pressure level estimates.

Proximity detection by proximity detector 4 may differ from near field sound detection by near field detector 3 because the signal characteristics of near speech may be very similar to ambient signals such as music and noise. Therefore, the proximity detector 4 must avoid false detection of near speech in order to achieve an acceptable user experience. Therefore, the music detector 9 can be used to disable proximity detection as long as there is music in the background. Likewise, the proximity detector 4 may be deactivated as long as the background noise sound pressure level is above a certain threshold. The background noise threshold can be determined a priori such that the probability of false detection below the threshold sound pressure level is very low. FIG. 6 illustrates exemplary fusion logic for detecting close sounds (eg, speech) according to an embodiment of the disclosure. In addition, there may be many sources of ambient noise that produce transient acoustic stimuli. These types of noise can be falsely detected as speech signals by speech detectors. To reduce the possibility of false detections, the Spectral Flatness Measure (SFM) statistic from the music detector 9 can be used to distinguish speech from transient noise. For example, SFM can be tracked over a period of time, and the difference between the maximum and minimum SFM values over the same period of time can be calculated, defined as sfmSwing. The value of sfmSwing may generally be small for transient noise signals because the spectral components of these signals are broadband in nature and they level off over short time intervals (300ms-500ms). The value of sfmSwing may be higher for speech signals, since the spectral content of speech signals may vary faster than instantaneous signals. As shown in Figure 6, proximity sounds (e.g., voices) can be detected when all of the following criteria are met:

No music detected in the background (block 20);

Direction of Arrival estimated DOA is within the acceptance angle of close sounds (block 21);

The maximum normalized cross-correlation statistic normMaxCorr is greater than the threshold normMaxCorrThres2 (block 22);

The background noise sound pressure level noiseLevel is lower than the threshold noiseLevelTh (block 23);

Proximity speech activity is detected, as indicated by the signal proxSpeechDet (block 19);

The SFM change statistic sfmSwing is greater than the threshold sfmSwingTh (block 37);

The interference to near-field desired signal ratio idr is greater than the threshold idrThres2 (block 40);

• The inter-microphone sound pressure level difference statistic imd approaches 0 dB (block 43).

In some embodiments, the music detector 9 used to detect the presence of background music may be implemented using the music detector taught in US Patent No. 8,126,706. According to an embodiment of the present disclosure, another embodiment of the proximity speech detector is shown in FIG. 7 . According to the present embodiment, a close voice can be detected if the following conditions are satisfied.

The interference to near-field desired signal ratio idr is greater than the threshold idrThres2 (block 39);

· Proximity voice activity is detected (block 27);

The maximum normalized cross-correlation statistic normMaxCorr is greater than the threshold normMaxCorrThres3 (block 28);

Direction of Arrival estimated DOA is within the acceptance angle of near sounds (block 29);

No music detected in the background (block 30);

• Presence of low or medium sound pressure level background noise or absence of background noise (block 31 ). This condition is verified by comparing the estimated background noise sound pressure level with the threshold noiseLevelThLo. If low noise sound pressure levels are detected, the following two conditions are also tested to confirm the presence of close speech:

The SFM change statistic sfmSwing is greater than the threshold sfmSwingTh (block 38);

• The inter-microphone sound pressure level difference statistic imd approaches 0 dB (block 44).

If the above-mentioned background noise sound pressure level condition is not met at block 31, the following conditions may represent close speech to increase the detection rate of close speech without increasing the occurrence of false alarms (e.g., due to background noise conditions):

• There is stationary background noise (block 32). Stationary background noise can be detected by calculating the peak-to-rms ratio of the SFM generated by the music detector (block 9) over a period of time. Specifically, if the above ratio is high, non-stationary noise may be present, since the spectral flatness measure of non-stationary noise tends to change faster than stationary noise;

• There is a high noise sound pressure level (block 32). A high noise condition may be detected if the estimated background noise is greater than a threshold noiseLevelLo and less than a threshold noiseLevelHi. If the above stationary noise and direction of arrival conditions are not met at block 32, the presence of the following set of two conditions may indicate the presence of close speech:

• Proximity talkers with intimate conversations (block 33). When the maximum normalized cross-correlation statistic normMaxCorr is greater than the threshold normMaxCorrThres4 (threshold normMaxCorrThres4 may be greater than normMaxCorrThres3 to indicate the presence of the intimate interlocutor), the close chatter of the intimate conversation can be detected;

• Presence of low or medium or high sound pressure level background noise or absence of background noise (block 34). This condition may be detected if the estimated background noise sound pressure level is less than a threshold noiseLevelThHi.

If the DOA condition above is not met at block 29, then the presence of the following conditions may indicate close speech:

No music is present (block 35);

• Proximity talkers with intimate conversations (block 33). When the maximum normalized cross-correlation statistic normMaxCorr is greater than the threshold normMaxCorrThres4 (threshold normMaxCorrThres4 may be greater than normMaxCorrThres3 to indicate the presence of the intimate interlocutor), the close chatter of the intimate conversation can be detected;

• Presence of low or medium or high sound pressure level background noise or absence of background noise (block 34). This condition may be detected if the estimated background noise sound pressure level is less than a threshold noiseLevelThHi.

The tonal alarm detector 5 may be configured to detect tonal alarm signals, wherein such alarm signals also have a narrow acoustic bandwidth (eg siren, buzzer). In some embodiments, the pitch of the ambient sound can be detected by dividing the time-domain signal into a plurality of sub-bands through time-frequency transformation, and the signal shown in FIG. 6 as generated by the music detector 9 can be calculated in each sub-band Spectral flatness measure for sfm[]. The spectral flatness measure sfm[] can be estimated for all subbands, and a tone alarm can be detected if the spectrum is flat in most but not all of the subbands. Also, in playback management systems, there may be no need to detect far-field alarm signals. Therefore, the near-field spatial statistics 8 of FIG. 3 can be used to distinguish far-field warning signals from near-field signals. 8 illustrates exemplary fusion logic for detecting a tone alarm event (eg, siren, buzzer) according to an embodiment of the disclosure. As shown in Figure 8, a tone alarm event can be detected when all of the following criteria are met:

Direction of arrival estimated DOA is within the acceptance angle of the alarm signal (block 24);

The maximum normalized cross-correlation statistic normMaxCorr is greater than the threshold normMaxCorrThres5 (block 25);

• The spectral flatness measure sfm[] indicates that the noise spectrum is flat in most but not all subbands (block 26).

Indeed, transient audio event detection by near field detector 3, proximity detector 4 and tone alarm detector 5 as shown in Figs. 5, 6, 7 and 8 may indicate false audio events. Therefore, it may be advisable to verify the transient audio event detection signal before passing the event detection signal to the playback control section 6 . 9 illustrates an exemplary timing diagram illustrating delay and hysteresis logic that may be applied to a transient audio event detection signal to generate a verified audio event signal, according to an embodiment of the disclosure. As shown in FIG. 9, in response to a momentary detection of an audio event (e.g., near field sound, near sound, tone alarm event) for at least a predetermined time, delay logic may generate a verified audio event signal, while hysteresis logic may continue to use The verified audio event signal is valid until the momentary detection of the audio event has ceased within the second predetermined time.

The following pseudocode may demonstrate the application of delay and hysteresis logic to reduce false detection of audio events according to an embodiment of the present disclosure.

/*If the instant. detect is true, increment the hold off counter and reset the hang over counter*/

If(instDet==TRUE)

{

holdOffCntr=holdOffCntr+1;

hangOverCntr = 0;

}

/*If the instant. detect is false, increment the hang over counter and reset the hold off counter*/

else

{

hangOverCntr=hangOverCntr+1;

holdOffCntr = 0;

}

/******************

*Hold-off Logic*

*******************/

/*Valid detect will transition to true state if the instant.detect is continuously true for certain time and the previous valid detect is false*/if(holdOffCntr>holdOffThres&&validDet==FALSE)

{

validDet = TRUE;

holdOffCntr = 0;

hangOverCntr = 0;

}

/******************

*Hang-Over Logic*

*******************/

/*Valid NF detect will transition to false state if the instant.NFdetect is continuously false for certain time and the previous valid NFdetect is true*/

If(hangOverCntr>hangOverThres&&validDet==TRUE)

{

validDet = FALSE;

holdOffCntr = 0;

hangOverCntr = 0;

}

Authenticated events can be further validated before generating the playback mode switch control. For example, the following pseudocode can be demonstrated for use in talk mode (e.g., where the audio information reproduced to the output audio transducer 51 can be modified in response to an audio event) and normal playback mode (e.g., where the output audio transducer 51 is not modified). The application of delay and hysteresis logic to switch gracefully between the audio information reproduced by the device 51).

/************************************

*Conversational Mode Enter Logic*

*************************************/

/*Increment the time to enter conversational mode counter if the event detect is true and the mode is not in the conversational mode. If the counter exceeds the threshold, switch to conversational mode and reset the counters. Note that the event detect need not be true contiguously. */if(convModeEn==FALSE&&validDet==TRUE)

{

timeToEnterConvModeCntr=timeToEnterConvModeCntr+1;

if(timeToEnterConvModeCntr>timeToEnterConvModeThres)

{

convModeEn = TRUE;

timeToEnterConvModeCntr = 0;

timeToExitConvModeCntr = 0;

}

}

/************************************

*Conversational Mode Exit Logic*

*************************************/

/*Increment the time to exit conversational mode counter if the event detect is false and the mode is in the conversational mode. If the counter exceeds the threshold, switch to normal mode and reset the counters. Note that the event detect must be false contiguously.*/

if(convModeEn==TRUE&&validDet==FALSE)

{

timeToExitConvModeCntr++;

if(timeToExitConvModeCntr>timeToExitConvModeThres)

{

convModeEn = FALSE;

timeToEnterConvModeCntr = 0;

timeToExitConvModeCntr = 0;

}

}

else

{

timeToExitConvModeCntr = 0;

}

Figure 10 illustrates different audio event detectors with delay and hysteresis logic according to an embodiment of the disclosure. The delay period and/or hysteresis period of each detector may be set differently. Additionally, in some embodiments, playback management may be controlled differently based on the type of event detected. In these and other embodiments, as shown in FIG. 9, the playback gain (and thus attenuation of the audio information reproduced at the output audio transducer 51) may be attenuated whenever one or more of the audio events are detected. . In these and other embodiments, to provide smooth gain transitions, the playback gain may be smoothed using a first order exponential averaging filter represented by the following pseudocode:

if(convModeEn==TRUE)

{

playBackGain=(1-alpha)*convModeGain+alpha*playBackGain

}

else

{

playBackGain=(1-beta)*normalModeGain+beta*playBackGain

}

The smoothing parameters α and β can be set to different values to adjust the gain slope.

It should be understood that various operations described herein, especially with reference to the accompanying drawings, may be implemented by other circuits or other hardware components, especially by those skilled in the art having benefited from the present disclosure. The order in which the various operations of a given method are performed may be changed, and various elements of the systems shown herein may be added, reordered, combined, omitted, modified, etc. The present disclosure is intended to embrace all such modifications and changes, and thus, the above description should be considered as illustrative rather than restrictive.

Likewise, while this disclosure refers to specific embodiments, certain modifications and changes may be made to these embodiments without departing from the scope of the present disclosure. Furthermore, any benefits, advantages or solutions described herein with respect to specific embodiments are not intended to be construed as critical, required or essential features or elements.

Further embodiments will likewise be apparent to persons of ordinary skill in the art having the benefit of this disclosure, and such embodiments should be considered to be included herein.

Claims (68)

1. A method for processing audio information in an audio device, the method comprising: reproducing audio information by generating an audio output signal for delivery to at least one transducer of said audio device; receiving at least one input signal representing ambient sound external to the audio device; detecting near-field sound among ambient sounds based on the at least one input signal; In response to detecting near-field sound, a characteristic of audio information reproduced to the at least one transducer is modified. 2. The method of claim 1 , further comprising determining the direction of the ambient sound based on the at least one input signal, and modifying the direction of the ambient sound in response to the direction of the ambient sound indicating that the ambient sound is from a user of the audio device. A characteristic of the audio information reproduced by the at least one transducer. 3. The method of claim 1 , further comprising determining the direction of the ambient sound based on the at least one input signal, and modifying the direction of the ambient sound in response to the direction of the ambient sound indicating that the ambient sound is from a user's voice of the audio device. A characteristic of the audio information reproduced by the at least one transducer. 4. The method of claim 1, wherein modifying a characteristic of the audio information comprises attenuating the audio information. 5. The method of claim 1, further comprising modifying a characteristic of the audio information in response to detecting near-field sound for at least a predetermined time. 6. The method of claim 5, further comprising: detecting the absence of near-field sound among ambient sounds based on the at least one input signal; In response to the absence of near-field sound for at least a second predetermined time, modifying characteristics of the audio information reproduced to the at least one transducer ceases. 7. The method of claim 1, further comprising: In addition to detecting near-field sounds, detecting ambient sounds other than near-field sounds in the ambient sounds from the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to detecting ambient sound. 8. The method according to claim 7, further comprising determining the direction of the surrounding sound based on the at least one input signal, and modifying the direction of the surrounding sound in response to the direction of the surrounding sound indicating that the surrounding sound is a sound other than near-field sound. A characteristic of the audio information reproduced by the at least one transducer. 9. The method of claim 7, further comprising: detecting whether the ambient sound includes background noise based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to detecting background noise in the ambient sound. 10. The method of claim 7, further comprising: detecting whether the ambient sound includes a tone alarm based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to detecting the tonal alert in the ambient sound. 11. The method of claim 10, wherein detecting a tone alarm in ambient sound comprises: detecting the direction of ambient sound from the at least one input signal; detecting a measure of spectral flatness of ambient sound from said at least one input signal; detecting near-field spatial statistics of ambient sound from the at least one input signal; Tonal alarms are detected based on the direction of surrounding sounds, the presence or absence of background noise, and near-field spatial statistics. 12. The method of claim 11, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include correlations between the first microphone signal and the second microphone signal. 13. The method of claim 11, wherein detecting the direction of ambient sound comprises determining whether the direction of ambient sound is within an acceptance angle for near-field sound. 14. The method of claim 11, wherein detecting near-field spatial statistics of ambient sound comprises detecting whether a normalized cross-correlation statistic is greater than a threshold. 15. The method of claim 11, wherein detecting the measure of spectral flatness of the ambient sound comprises detecting whether the noise spectrum is flat in most subbands of the ambient sound but not all subbands of the ambient sound. 16. The method of claim 1, wherein detecting near-field sounds among ambient sounds comprises: detecting the direction of ambient sound from the at least one input signal; detecting the presence of voice in ambient sound from the at least one input signal; detecting near-field spatial statistics of ambient sound from the at least one input signal; Near-field sounds are detected based on the direction of surrounding sounds, the presence or absence of voice, and near-field spatial statistics. 17. The method of claim 16, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include correlations between the first microphone signal and the second microphone signal. 18. The method of claim 16, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an interference-to-signal ratio associated with near-field sound. 19. The method of claim 16, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an inter-microphone sound pressure level difference between the first microphone signal and the second microphone signal. 20. The method of claim 16, wherein detecting the direction of ambient sound comprises determining whether the direction of ambient sound is within an acceptance angle for near-field sound. 21. The method of claim 16, wherein detecting near-field spatial statistics of ambient sound comprises: Detecting whether the standardized cross-correlation statistic is greater than a first threshold; Detecting whether the ratio of the interference to the near-field desired signal is smaller than a second threshold; Detecting whether the sound pressure level difference between the microphones is greater than a third threshold. 22. The method of claim 21, wherein the second threshold is adjusted based on an estimate of background noise in ambient sounds. 23. The method of claim 21, wherein the third threshold is adjusted based on an estimate of background noise in ambient sounds. 24. The method of claim 1, further comprising: detecting the direction of ambient sound from the at least one input signal; detecting the presence of background noise in ambient sound from the at least one input signal; detecting the presence of close speech in ambient sound from the at least one input signal; detecting the volume of ambient sound from the at least one input signal; detecting near-field spatial statistics of ambient sound from the at least one input signal; Detecting the presence of audio events, including near-field sound events, based on the direction of surrounding sounds, the presence or absence of background noise, the presence or absence of speech, volume, and near-field spatial statistics; A characteristic of the audio information reproduced to the at least one transducer is modified in response to the presence of the audio event. 25. The method of claim 24, further comprising: Detect changes in the spectral content of surrounding sounds; The presence of audio events, including close sound events, is detected based on the direction of ambient sounds, presence or absence of background noise, presence or absence of speech, volume, near-field spatial statistics, and spectral content of ambient sounds. 26. The method of claim 25, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include correlations between the first microphone signal and the second microphone signal. 27. The method of claim 25, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an interference-to-signal ratio associated with near-field sound. 28. The method of claim 25, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an inter-microphone sound pressure level difference between the first microphone signal and the second microphone signal. 29. The method of claim 25, wherein detecting the presence of close speech in ambient sound comprises detecting stationary background noise. 30. The method of claim 25, wherein detecting the presence of close speech in ambient sounds comprises detecting speech from a close talker of an intimate conversation. 31. The method of claim 25, wherein detecting the presence of close speech in ambient sound comprises detecting a measure of spectral flatness of the ambient sound from the at least one input signal, wherein detecting the measure of spectral flatness of the ambient sound This includes detecting changes in the spectral content of ambient sounds. 32. An integrated circuit for implementing at least a portion of an audio device, the integrated circuit comprising: an audio output configured to reproduce audio information by generating an audio output signal for delivery to at least one transducer of said audio device; a microphone input configured to receive an input signal representative of ambient sound external to the audio device; processor, configured to: detecting near-field sound from ambient sound based on the input signal; A characteristic of the audio information is modified in response to detecting near-field sound. 33. The integrated circuit of claim 32, the processor further configured to: determining the direction of ambient sound based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to the direction of the ambient sound indicating that the ambient sound is from a user of the audio device. 34. The integrated circuit of claim 32, the processor further configured to: determining the direction of ambient sound based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to the direction of the ambient sound indicating that the ambient sound is voice from a user of the audio device. 35. The integrated circuit of claim 32, wherein modifying a characteristic of the audio information comprises attenuating the audio information. 36. The integrated circuit of claim 32, the processor further configured to modify a characteristic of the audio information in response to detecting near-field sound for at least a predetermined time. 37. The integrated circuit of claim 36, the processor further configured to: detecting the absence of near-field sound among ambient sounds based on the at least one input signal; In response to the absence of near-field sound for at least a second predetermined time, modifying characteristics of the audio information reproduced to the at least one transducer ceases. 38. The integrated circuit of claim 36, the processor further configured to: In addition to detecting near-field sounds, detecting ambient sounds other than near-field sounds in the ambient sounds from the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to detecting ambient sound. 39. The integrated circuit of claim 38, the processor further configured to: determining the direction of ambient sound based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to the direction of the ambient sound indicating that the ambient sound is a sound other than near-field sound. 40. The integrated circuit of claim 38, the processor further configured to: detecting whether the ambient sound includes background noise based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to detecting background noise in the ambient sound. 41. The integrated circuit of claim 38, the processor further configured to: detecting whether the ambient sound includes a tone alarm based on the at least one input signal; A characteristic of the audio information reproduced to the at least one transducer is modified in response to detecting the tonal alert in the ambient sound. 42. The integrated circuit of claim 41 , wherein detecting a tone alarm in ambient sound comprises: detecting the direction of ambient sound from the at least one input signal; detecting a measure of spectral flatness of ambient sound from said at least one input signal; detecting near-field spatial statistics of ambient sound from the at least one input signal; Tonal alarms are detected based on the direction of surrounding sounds, the presence or absence of background noise, and near-field spatial statistics. 43. The integrated circuit of claim 41, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include correlations between the first microphone signal and the second microphone signal. 44. The integrated circuit of claim 42, wherein detecting the direction of ambient sound includes determining whether the direction of ambient sound is within an acceptance angle for near-field sound. 45. The integrated circuit of claim 42, wherein detecting near-field spatial statistics of ambient sound includes detecting whether a normalized cross-correlation statistic is greater than a threshold. 46. The integrated circuit of claim 42, wherein detecting the measure of spectral flatness of the ambient sound comprises detecting whether the noise spectrum is flat in most subbands of the ambient sound but not all subbands of the ambient sound. 47. The integrated circuit of claim 32, wherein detecting near-field sound in ambient sound comprises: detecting the direction of ambient sound from the at least one input signal; detecting the presence of voice in ambient sound from the at least one input signal; detecting near-field spatial statistics of ambient sound from the at least one input signal; Near-field sounds are detected based on the direction of surrounding sounds, the presence or absence of voice, and near-field spatial statistics. 48. The integrated circuit of claim 47, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include correlations between the first microphone signal and the second microphone signal. 49. The integrated circuit of claim 47, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an interference-to-signal ratio associated with near-field sound. 50. The method of claim 47, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an inter-microphone sound pressure level difference between the first microphone signal and the second microphone signal. 51. The integrated circuit of claim 47, wherein detecting the direction of ambient sound includes determining whether the direction of ambient sound is within an acceptance angle for near-field sound. 52. The integrated circuit of claim 47, wherein detecting near-field spatial statistics of ambient sound comprises: Detecting whether the standardized cross-correlation statistic is greater than a first threshold; Detecting whether the ratio of the interference to the near-field desired signal is smaller than a second threshold; Detecting whether the sound pressure level difference between the microphones is greater than a third threshold. 53. The integrated circuit of claim 52, wherein the second threshold is adjusted based on an estimate of background noise in ambient sound. 54. The integrated circuit of claim 52, wherein the second threshold is adjusted based on an estimate of background noise in ambient sound. 55. The integrated circuit of claim 32, the processor further configured to: detecting the direction of ambient sound from the at least one input signal; detecting the presence of background noise in ambient sound from the at least one input signal; detecting the presence of close speech in ambient sound from the at least one input signal; detecting the volume of ambient sound from the at least one input signal; detecting near-field spatial statistics of ambient sound from the at least one input signal; Detecting the presence of audio events, including near-field sound events, based on the direction of surrounding sounds, the presence or absence of background noise, the presence or absence of speech, volume, and near-field spatial statistics; A characteristic of the audio information reproduced to the at least one transducer is modified in response to the presence of the audio event. 56. The integrated circuit of claim 32, the processor further configured to: Detect changes in the spectral content of surrounding sounds; The presence of audio events, including close sound events, is detected based on the direction of ambient sounds, presence or absence of background noise, presence or absence of speech, volume, near-field spatial statistics, and spectral content of ambient sounds. 57. The integrated circuit of claim 56, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include correlations between the first microphone signal and the second microphone signal. 58. The integrated circuit of claim 56, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an interference-to-signal ratio associated with near-field sound. 59. The integrated circuit of claim 56, wherein: The at least one input signal comprises a first microphone signal representing ambient sound at the first microphone and a second microphone signal representing ambient sound at the second microphone; The near-field spatial statistics include an inter-microphone sound pressure level difference between the first microphone signal and the second microphone signal. 60. The integrated circuit of claim 56, wherein detecting the presence of close speech in ambient sound comprises detecting stationary background noise. 61. The integrated circuit of claim 56, wherein detecting the presence of close speech in ambient sound comprises detecting speech from a close talker of an intimate conversation. 62. The integrated circuit of claim 56, wherein detecting the presence of close speech in ambient sound comprises detecting a measure of spectral flatness of the ambient sound from the at least one input signal, wherein detecting the spectral flatness of the ambient sound Measurements include detecting changes in the spectral content of ambient sounds. 63. A method for processing audio information in an audio device, the method comprising: reproducing audio information by generating an audio output signal for delivery to at least one transducer of said audio device; receiving at least one input signal representing ambient sound external to the audio device; detecting an audio event based on the at least one input signal; In response to detecting an audio event for at least a predetermined time, a characteristic of the audio information reproduced to the at least one transducer is modified. 64. The method of claim 63, further comprising ceasing to modify characteristics of audio information reproduced to the at least one transducer in response to the absence of an audio event for at least a second predetermined time. 65. The method of claim 63, wherein the audio event comprises at least one of a near-field event, a near-range event, and an alarm event. 66. An integrated circuit for implementing at least a portion of an audio device, the integrated circuit comprising: an audio output configured to reproduce audio information by generating an audio output signal for delivery to at least one transducer of said audio device; a microphone input configured to receive an input signal representative of ambient sound external to the audio device; processor, configured to: detecting an audio event based on the input signal; In response to detecting an audio event for at least a predetermined time, a characteristic of the audio information reproduced to the at least one transducer is modified. 67. The integrated circuit of claim 66 , the processor further configured to cease modifying characteristics of audio information reproduced to the at least one transducer in response to an absence of an audio event for at least a second predetermined time. . 68. The integrated circuit of claim 66, wherein the audio event comprises at least one of a near-field event, a near-range event, and an alarm event.