Audio on-demand communication system

• US 8,706,903 B2
• Filed: 12/01/2011
• Issued: 04/22/2014
• Est. Priority Date: 11/30/1994
• Status: Expired due to Fees

• Current Assignee: Intel Corporation
• Original Assignee: Intel Corporation
• Inventors: Glaser, Robert D., O'Brien, Mark, Boutell, Thomas B., Goldberg, Randy Glen
• Primary Examiner(s): Hussain, Tauqir
• Application Number:
  US13/309,454
• Publication Number:
  US 20120148064A1
• Time in Patent Office: 873 Days
• Field of Search: 709/236
• US Class Current: 709/236
• CPC Class Codes: H04H 20/28 Arrangements for simultaneous broadcast of plural pieces of information
• CPC Class Codes: H04H 20/30 by a single channel
• CPC Class Codes: H04H 20/40 Arrangements for broadcast specially adapted for accumulation-type receivers
• CPC Class Codes: H04H 20/46 specially adapted for broadcast systems covered by groups H04H20/53-H04H20/95
• CPC Class Codes: H04H 20/82 using signals not modulated onto a carrier
• CPC Class Codes: H04H 20/83 not sharing the network with any other service
• CPC Class Codes: H04H 60/27 Arrangements for recording or accumulating broadcast information or broadcast-related information
• CPC Class Codes: H04H 60/51 of receiving stations
• CPC Class Codes: H04H 60/73 using meta-information

Abstract

An audio-on-demand communication system provides real-time playback of audio data transferred via telephone lines or other communication links. One or more audio servers include memory banks which store compressed audio data. At the request of a user at a subscriber PC, an audio server transmits the compressed audio data over the communication link to the subscriber PC. The subscriber PC receives and decompresses the transmitted audio data in less than real-time using only the processing power of the CPU within the subscriber PC. According to one aspect of the present invention, high quality audio data compressed according to lossless compression techniques is transmitted together with normal quality audio data. According to another aspect of the present invention, metadata, or extra data, such as text, captions, still images, etc., is transmitted with audio data and is simultaneously displayed with corresponding audio data.

Claims

1. An apparatus comprising:

a receiving circuit to;

obtain a normal quality audio stream from one or more servers and store the normal quality audio stream in a normal quality buffer, and

obtain at least one higher quality audio segment from the one or more servers and store the at least one higher quality audio segment in a high quality buffer,

wherein the higher quality audio segment is a less digitally compressed version of a corresponding segment of the normal quality audio stream, and

wherein the higher quality audio segment is associated with a time stamp for synchronizing presentation of the higher quality audio segment during playback of the normal quality audio stream;

a playback circuit to output the normal quality audio stream from the normal quality buffer; and

a processor to automatically present the higher quality audio segment from the high quality buffer to the playback circuit in place of the corresponding segment of the normal quality audio stream from the normal quality buffer at a time specified by the time stamp.

2. The apparatus of claim 1, wherein the higher quality audio segment is interleaved with the normal quality audio stream.

3. The apparatus of claim 2, wherein the higher quality audio segment is interleaved with a first portion of the normal quality audio stream, the first portion being prior to the time within the normal quality audio stream specified by the time stamp.

4. The apparatus of claim 2, wherein the higher quality audio segment is interleaved with the normal quality audio stream according a determined ratio.

5. The apparatus of claim 3, wherein the normal quality audio stream and the higher quality audio segment are subdivided into blocks, and wherein the ratio comprises at least two normal quality audio stream blocks for each higher quality audio segment block.

6. The apparatus of claim 1, wherein the processor is to monitor a level of the normal quality buffer and transmit a first signal to the one or more servers if the buffer level reaches at least a first threshold, the first signal being configured to cause the one or more servers to transmit the higher quality audio segment with the normal quality audio stream.

7. The apparatus of claim 6, wherein the processor is to transmit a second signal to the one or more servers if the buffer level drops below a second threshold, the second signal being configured to cause the one or more servers to discontinue transmitting the higher quality audio segment with the normal quality audio stream.

8. The apparatus of claim 7, wherein the first threshold is equal to the second threshold.

9. The apparatus of claim 1, wherein at least one buffer is to store a determined amount of the normal quality audio stream and/or the higher quality audio segment before the playback circuit is to begin outputting the normal quality audio stream.

10. The apparatus of claim 1, wherein the processor is to regulate a rate at which the normal quality audio stream and/or the higher quality audio segment is received from the one or more servers.

11. The apparatus of claim 10, wherein the processor is to regulate the rate at which the normal quality audio stream and/or the higher quality audio segment is received by selectively transmitting a signal to the one or more servers in response to at least one marker being identified in the normal quality audio stream and/or the higher quality audio segment.

12. The apparatus of claim 11, wherein the at least one marker is interleaved with the normal quality audio stream and/or the higher quality audio segment.

13. The apparatus of claim 11, wherein the signal is transmitted upon encountering the at least one marker if a determined amount of space is available in at least one buffer.

14. The apparatus of claim 11, wherein failure to acknowledge the at least one marker is to cause the one or more servers to discontinue transmitting the normal quality audio stream and/or the higher quality audio segment.

15. The apparatus of claim 1, wherein, in response to receiving a seek command identifying a portion of the normal quality audio stream to be played back, the processor is to transmit a first signal to the one or more servers, the first signal being configured to cause the one or more servers to begin transmitting the higher quality audio segment related to the identified portion of the normal quality audio stream.

16. The apparatus of claim 1, wherein the normal quality audio stream is transmitted from a first server and the higher quality audio segment is transmitted from a second server.

17. The apparatus of claim 1, wherein the one or more servers are selected based on geographic location.

18. A method comprising:

receiving at a client device a normal quality audio stream sent by one more servers and storing the normal quality audio stream in a normal quality buffer;

receiving at least one higher quality audio segment sent by the one or more servers and storing the at least one higher quality audio segment in a high quality buffer, wherein the higher quality audio segment is a less digitally compressed version of a corresponding segment of the normal quality audio stream, and wherein the higher quality audio segment is associated with at least one time stamp for synchronizing presentation of the higher quality audio segment during playback of the normal quality audio stream; and

while outputting the normal quality audio stream from the normal quality buffer, automatically presenting the higher quality audio segment from the high quality buffer at a time within the normal quality audio stream specified by the time stamp.

19. The method of claim 18, further comprising:

monitoring a level of the normal quality buffer; and

transmitting a first signal to the one or more servers if the buffer level reaches at least a first threshold, the first signal being configured to cause the one or more servers to transmit the higher quality audio segment with the normal quality audio stream.

20. The method of claim 19, further comprising transmitting a second signal to the one or more servers if the buffer level drops below a second threshold, the second signal being configured to cause the one or more servers to discontinue transmitting the higher quality audio segment with the normal quality audio stream.

21. The method of claim 20, wherein the first threshold is equal to the second threshold.

22. The method of claim 18, further comprising buffering a determined amount of the normal quality audio stream and/or the higher quality audio segment before outputting the normal quality audio stream.

23. The method of claim 18, further comprising:

receiving a seek command identifying a portion of the normal quality audio stream to be played back; and

transmitting a first signal to the one or more servers, the first signal being configured to cause the one or more servers to begin transmitting the higher quality audio segment related to the identified portion of the normal quality audio stream.

24. The method of claim 18, further comprising regulating a rate at which the normal quality audio stream and/or the higher quality audio segment is received from the one or more servers by selectively transmitting a signal to the one or more servers in response to at least one marker being identified in the normal quality audio stream, and wherein failure to acknowledge the at least one marker is to cause the one or more servers to discontinue transmitting the normal quality audio stream and/or the higher quality audio segment.