System and method for providing real-time communication of high quality audio

• US 20080059197A1
• Filed: 08/29/2006
• Published: 03/06/2008
• Est. Priority Date: 08/29/2006
• Status: Abandoned Application

• Current Assignee: Chartlogic Incorporated (Medsphere Systems Corporation)
• Original Assignee: Chartlogic Incorporated (Medsphere Systems Corporation)
• Inventors: Berry, Michael J., Jones, Gary A.
• Application Number:
  US11/512,021
• Publication Number:
  US 20080059197A1
• Time in Patent Office: Days
• US Class Current: 704/270.1
• CPC Class Codes: G10L 15/30 Distributed recognition, e.g. in client-server systems, for mobile phones or network applications

Abstract

A system and method for providing real-time communication of high-quality audio is provided. The system contains a series of audio devices and a series of client devices, where each client device is in communication with an audio device, and each client device is capable of converting analog signals received from an audio device, into digital data. In addition, a series of server devices is provided where each service device is capable of communicating with one of the series of client devices via a network, and each server device is capable of converting digital data received from a client device, into analog signals. A series of server computers is provided, each having a sound card. A connection between a server device and a server computer resulting in analog signals from the server device being directly received by the sound card located within the server computer.

Claims

1. A system for providing real-time communication of high-quality audio, comprising:

an audio device;

at least one client device in communication with said audio device, said client device being capable of converting analog signals received from said audio device, into digital data;

a network for allowing communication within said system;

at least one server device capable of communicating with said client device via said network, said server device being capable of converting digital data received from said client device, into analog signals; and

a server in communication with said server device and said network, said server having a sound card, a connection between said server device and said server resulting in analog signals from said server device being directly received by said sound card located within said server.

2. The system of claim 1, further comprising at least one client computer capable of communicating with said server via said network, said client computer containing a means for allowing a user of said client computer to provide user information and a screen for displaying text received from said server.

3. The system of claim 2, wherein said server further comprises automatic speech recognition (ASR) software, said ASR software being capable of being used to transcribe said received analog signals, with respect to a user voice file, into said text for transmission to said client computer via said network.

4. The system of claim 3, wherein said user voice file is located within a storage device of said server.

5. The system of claim 3, where said user voice file is located remote from said server and retrieved by said server for transcription of said analog signals.

6. The system of claim 3, wherein said server and said client computer each further comprise a memory and a processor, wherein said memory of said server and said memory of said client computer further comprise connection software stored therein, and wherein said processor of said server and said processor of said computer are configured by said connection software to perform the steps of:

defining an audio signal transmission path from said client device to said server device; and

specifying relationships between said client computer and said client device resulting in said transcribed text associated with said digital data transmitted by said client device, being transmitted to said client computer.

7. The system of claim 3, where said ASR software is also capable of determining identity of a user that derived the received audio signals.

8. The system of claim 3, wherein said server further comprises a second ASR software stored within said server that is capable of determining identity of a user that derived the received audio signals.

9. The system of claim 1, wherein said client device further comprises:

means for communicating with said audio device;

an encoder/decoder (CODEC) connected to said means for communicating with said audio device, said CODEC capable of converting analog signals received via the means for communicating into digital data;

a digital signal processor connected to said CODEC, said digital signal processor capable of performing data conversion of full-duplex serial digital audio streams; and

a device server connected to said digital signal processor, said device server capable of converting asynchronous serial data received from the digital signal processor to streaming Internet Protocol (IP) packets.

10. The system of claim 9, wherein said digital signal processor (DSP) further comprises a buffer, and wherein said DSP of said client device is capable of performing the steps of:

when an audio sample is received by said DSP, determining when said buffer is more than a percentage X full and less than a percentage Y full;

if said buffer is more than percentage X full, said DSP setting a flag to delete an audio sample from audio samples that are to be provided to said CODEC;

if said buffer is less than percentage Y full, said DSP setting said flag to add an audio sample to said audio samples that are going to be provided to said CODEC; and

if said buffer is not more than percentage X full and not less than percentage Y full, said DSP leaving a state of said flag unchanged.

11. The system of claim 10, wherein said DSP of said client device is also capable of performing the step of, after a predefined number of audio samples have been received by said DSP, said DSP being ready to add or delete an audio sample, and when there is a zero-crossing point in audio signals received, said DSP adding or deleting audio samples in accordance with said set flag.

12. The system of claim 9, wherein said digital signal processor (DSP) of said client device is capable of performing byte alignment, said byte alignment comprising the steps of:

inserting a predefined bit pattern into an audio sample stream being transmitted to said server device, said bit pattern representing that a next received audio sample will be either a low order audio sample or a high order audio sample;

transmitting said predefined bit pattern to said server device; and

adjusting byte alignment if necessary within said client device if a predefined bit pattern is received.

13. The system of claim 9, wherein said means for communicating with said audio device is a wireless communication device.

14. The system of claim 9, wherein said means for communicating with said audio device comprises a microphone jack and a speaker jack.

15. The system of claim 1, wherein said connection between said server device and said server is provided by a line in jack and a line out jack of said server device, and a line in jack and a line out jack of said server.

16. The system of claim 1, wherein said connection between said server device and said server is a wireless connection comprising a first wireless communication device located within said server device and a second wireless communication device located within said server, wherein said second wireless communication device is directly connected to said soundcard.

17. The system of claim 1, wherein said network is selected from the group consisting of a local area network and a wide area network.

18. The system of claim 1, wherein said server device further comprises:

means for communicating with said server;

a digital signal processor, said digital signal processor capable of performing data conversion of full-duplex serial digital audio streams;

a server device encoder/decoder (CODEC) connected to said means for communicating with said server and connected to said digital signal processor, said server device CODEC capable of converting digital data received via said digital signal processor into analog signals; and

a device server connected to said digital signal processor, said device server capable of converting streaming Internet Protocol packets of data received from said network, into asynchronous serial data.

19. The system of claim 18, wherein said digital signal processor (DSP) further comprises a buffer, and wherein said DSP of said server device is capable of performing the steps of:

when an audio sample is received by said DSP, determining when said buffer is more than a percentage X full and less than a percentage Y full;

if said buffer is more than percentage X full, said DSP setting a flag to delete an audio sample from audio samples that are to be provided to said CODEC, if said buffer is less than percentage Y full, said DSP setting said flag to add an audio sample to said audio samples that are going to be provided to said CODEC; and

if said buffer is not more than percentage X full and not less than percentage Y full, said DSP leaving a state of said flag unchanged.

20. The system of claim 19, wherein said DSP of said server device is also capable of performing the step of, after a predefined number of audio samples have been received by said DSP, said DSP being ready to add or delete an audio sample, and when there is a zero-crossing point in audio signals received, said DSP adding or deleting audio samples in accordance with said set flag.

21. The system of claim 18, wherein said digital signal processor (DSP) of said server device is capable of performing byte alignment, said byte alignment comprising the steps of:

inserting a predefined bit pattern into an audio sample stream being transmitted to said client device, said bit pattern representing that a next received audio sample will be either a low order audio sample or a high order audio sample;

transmitting said predefined bit pattern to said client device; and

adjusting byte alignment if necessary within said server device if a predefined bit pattern is received.

22. A system for providing real-time communication of high-quality audio, comprising:

a series of audio devices;

a series of client devices, each one of said client devices in communication with one of said audio devices, each one of said client devices being capable of converting analog signals received from one audio device of said series of audio device, into digital data;

a network for allowing communication within said system;

a series of server devices, each one of said service devices being capable of communicating with one of said series of client devices via said network, each one of said server devices being capable of converting digital data received from one client device of said series of client devices, into analog signals; and

a series of server computers, each of said server computers having a sound card, a connection between one of said series of server devices and one of said series of server computers resulting in analog signals from said one of said series of server devices being directly received by said sound card located within said one of said series of server computers.

23. The system of claim 22, further comprising a series of client computers, wherein each client computer within said series of client computers is capable of communicating with one server computer of said series of server computers via said network, said each client computer containing a means for allowing a user of said each client computer to provide user information and a screen for displaying text received from said one of said series of server computers.

24. The system of claim 23, further comprising a server connected to said network, wherein said server, each client computer within said series of client computers, and each server computer of said series of server computers, each further comprise a memory and a processor, wherein said memory of said server, said memory of said each client computer, and said memory of said each server computer further comprises connection software stored therein, and wherein said processor of said server, said processor of said each client computer, and said processor of said each server computer is configured by said connection software to perform the steps of:

defining an audio signal transmission path from one client device of said series of client devices to one server device of said series of server devices; and

specifying relationships between one client computer of said series of client computers and one client device of said series of client devices resulting in said transcribed text associated with said digital data transmitted by said one of said series of client devices, being transmitted to said one of said series of client computers.

25. The system of claim 24, wherein each server computer of said series of server computers further comprises automatic speech recognition (ASR) software, said ASR software being capable of transcribing said received analog signals, with respect to a user voice file, into said text for transmission to a client computer of said series of client computers that is associated with said one client device from which said analog signals were originally derived.

26. The system of claim 25, wherein said user voice file is located within a storage device of said one server computer.

27. The system of claim 25, wherein said user voice file is located remote from said one server computer and retrieved by said one server computer for transcription of said analog signals.

28. The system of claim 25, wherein said ASR software is also capable of determining identity of a user that derived the received analog signals.

29. The system of claim 25, where each of said server computers further comprises a second ASR software that is capable of determining identity of a user that derived the received analog signals.

30. The system of claim 22, wherein said each client device further comprises:

means for communicating with one of said series of audio devices;

an encoder/decoder (CODEC) connected to said means for communicating with said one of said audio devices, said CODEC capable of converting analog signals received via the means for communicating into digital data;

a digital signal processor connected to said CODEC, said digital signal processor capable of performing data conversion of full-duplex serial digital audio streams; and

a device server connected to said digital signal processor, said device server capable of converting asynchronous serial data received from the digital signal processor to streaming Internet Protocol (IP) packets.

31. The system of claim 30, wherein said digital signal processor (DSP) further comprises a buffer, and wherein said DSP of said client device is capable of performing the steps of:

when an audio sample is received by said DSP, determining when said buffer is more than a percentage X full and less than a percentage Y full;

if said buffer is more than percentage X full, said DSP setting a flag to delete an audio sample from audio samples that are to be provided to said CODEC;

if said buffer is less than percentage Y full, said DSP setting said flag to add an audio sample to said audio samples that are going to be provided to said CODEC; and

if said buffer is not more than percentage X full and not less than percentage Y full, said DSP leaving a state of said flag unchanged.

32. The system of claim 31, wherein said DSP of said client device is also capable of performing the step of, after a predefined number of audio samples have been received by said DSP, said DSP being ready to add or delete an audio sample, and when there is a zero-crossing point in audio signals received, said DSP adding or deleting audio samples in accordance with said set flag.

33. The system of claim 30, wherein said digital signal processor (DSP) of said client device is capable of performing byte alignment, said byte alignment comprising the steps of:

inserting a predefined bit pattern into an audio sample stream being transmitted to said server device, said bit pattern representing that a next received audio sample will be either a low order audio sample or a high order audio sample;

transmitting said predefined bit pattern to said server device; and

adjusting byte alignment if necessary within said client device if a predefined bit pattern is received.

34. The system of claim 30, wherein said means for communicating with said audio device is a wireless communication device.

35. The system of claim 30, wherein said means for communicating with said audio device comprises a microphone jack and a speaker jack.

36. The system of claim 22, wherein said connection between one of said series of server devices and one of said series of server computers is provided by a line in jack and a line out jack of said server device, and a line in jack and a line out jack of said server computer.

37. The system of claim 22, wherein said connection between one of said server devices and one of said server computers is a wireless connection comprising a first wireless communication device located within said one server device and a second wireless communication device located within said one server computer, wherein said second wireless communication device is directly connected to said soundcard.

38. The system of claim 22, wherein said network is selected from the group consisting of a local area network and a wide area network.

39. The system of claim 22, wherein said server device further comprises:

means for communicating with one of said series of server computers;

a digital signal processor, said digital signal processor capable of performing data conversion of full-duplex serial digital audio streams;

a server device encoder/decoder (CODEC) connected to said means for communicating with said one of said series of server computers and connected to said digital signal processor, said server device CODEC capable of converting digital data received via said digital signal processor into analog signals; and

a device server connected to said digital signal processor, said device server capable of converting streaming Internet Protocol packets of data received from said network, into asynchronous serial data.

40. The system of claim 39, wherein said digital signal processor (DSP) further comprises a buffer, and wherein said DSP of said server device is capable of performing the steps of:

when an audio sample is received by said DSP, determining when said buffer is more than a percentage X full and less than a percentage Y full;

if said buffer is more than percentage X full, said DSP setting a flag to delete an audio sample from audio samples that are to be provided to said CODEC;

if said buffer is less than percentage Y full, said DSP setting said flag to add an audio sample to said audio samples that are going to be provided to said CODEC; and

if said buffer is not more than percentage X full and not less than percentage Y full, said DSP leaving a state of said flag unchanged.

41. The system of claim 40, wherein said DSP of said server device is also capable of performing the step of, after a predefined number of audio samples have been received by said DSP, said DSP being ready to add or delete an audio sample, and when there is a zero-crossing point in audio signals received, said DSP adding or deleting audio samples in accordance with said set flag.

42. The system of claim 39, wherein said digital signal processor (DSP) of said server device is capable of performing byte alignment, said byte alignment comprising the steps of:

inserting a predefined bit pattern into an audio sample stream being transmitted to said client device, said bit pattern representing that a next received audio sample will be either a low order audio sample or a high order audio sample;

transmitting said predefined bit pattern to said client device; and

adjusting byte alignment if necessary within said server device if a predefined bit pattern is received.