Real-time audio buffering for telephone handsets

Telephonic communications – Substation or terminal circuitry

Reexamination Certificate

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Details

C455S334000

Reexamination Certificate

active

06771768

ABSTRACT:

FIELD OF THE INVENTION
This invention relates generally to the field of telephone handsets, and more particularly to handsets that provide loss-less telephone communications.
BACKGROUND OF THE INVENTION
There are brief moments when a user of a telephone handset is distracted from a telephone conversations. Examples include interruptions by nearby people, loud background noises, or just unintentional distractions. In many of these cases, the user intentionally removes the handset from his or her ear to perform some other task that requires the use of the hand holding the handset.
In the prior art, two techniques are generally used to provide the user with a capability to receive all of the information in a telephone call. In one method, the incoming call is placed on “hold.” This requires that the user notifies the caller to stop the conversation, until the hold is removed. However, in many cases, the user may not want to notify the caller of the need to do something else. The hold function also requires special equipment that may not be available with all handsets. In addition, if the call generated from a recording, placing the call on hold will result in loss of information. Also, in many case, the need for a “hold” cannot be anticipated, or is not realized until after the fact.
The other method uses a “speaker phone.” This approach has two problems. First, the quality of a speaker phone call is generally degraded. Second, with a speaker phone, the call is audible by all people within range of the speaker phone. This may not always be desired.
Therefore, there is a need for a method and apparatus that enables loss-less reception of telephone calls, even when the handset is temporarily removed from the ear so that the user cannot hear the telephone call.
SUMMARY OF THE INVENTION
A telephone handset includes an ear proximity sensor to dynamically buffer an incoming audio signals when the handset is removed from the ear. The buffer audio signal is then played back when the handset is returned to the ear. By dropping silent intervals and speeding up the playback of the buffered audio signal with a pitch-preserving process, the buffered audio signal is quickly resynchronized with the real-time incoming audio signal, at which point the buffering of the incoming audio signal stops. This loss-less buffering and instant play-back of the real-time incoming audio signal enables the user to hear all of the audio signal even when the handset has been put down, or the handset is removed from the ear due to any other user activities.
In order to preserve context with the interrupted telephone call, the play-back includes several seconds of content from before when the handset was removed from the ear.
In those cases when the user fails to hear a portion of the call, for whatever reason, the user can simply remove the handset from the ear, and an “instant” replay is effected so that the user can hear the missed portion of the call. If the missed portion of the call was for a longer period, the user can repeatedly remove the handset from the ear. Each time the handset is removed, play-back resumes successively further back in time, allowing the user to replay arbitrarily long segments, within memory limits.
More particularly, a telephone handset is configured to dynamically buffer a real-time audio signal. The handset includes a microphone and a speaker connected to a telephone network interface. A proximity sensor is located near the speaker of the handset. The proximity sensor is connected to a circuit for measuring a coupling between the handset and an ear of a user. A buffer continuously records the real-time audio signal. The recorded signal can than be played back after the handset is first removed from the ear of the user, and second re-applied to the ear of the user using a time-compressed audio signal, until the playback signal is resynchronized with the real-time audio signal.


REFERENCES:
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patent: 5828965 (1998-10-01), Brown et al.
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patent: 5982764 (1999-11-01), Palermo et al.
Techniques, Perception, and Applications of Time-Compressed Speech, Barry Arons, Proceedings of 1992 Conference, American Voice I/O Society.

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