Clock skew compensation for a jitter buffer

• US 20030169755A1
• Filed: 03/11/2002
• Published: 09/11/2003
• Est. Priority Date: 03/11/2002
• Status: Active Grant

• Current Assignee: Synaptics Incorporated
• Original Assignee: GlobespanVirata, Inc. (Synaptics Incorporated)
• Inventors: Ternovsky, Igor
• Granted Patent: US 7,263,109 B2
• Time in Patent Office: Days
• US Class Current: 370/412
• CPC Class Codes: H04L 12/6418 Hybrid transport
• CPC Class Codes: H04L 2012/6481 Speech, voice
• CPC Class Codes: H04L 2012/6489 Buffer Management, Threshold setting, Scheduling, Shaping

Abstract

A method and system to minimize the potential of jitter buffer underflow/overflow resulting from a difference in sampling rates of an audio encoder and an audio decoder are disclosed herein. The difference in sampling rates, or clock skew, can be determined from a difference between an actual amount of data stored in a jitter buffer and the desired, or threshold, amount. A subset of packets from a sequence of packets output to the audio decoder can be altered to compensate for the clock skew, whereby the amount of data associated with the subset of packets is decreased when the sampling rate of the encoder is greater than the sampling rate of the decoder, and the amount of data is increased when the sampling rate of the encoder is less than the sampling rate of the decoder. The present invention finds particular advantage in providing audio data via a packet-switched network.

Claims

1. A method for processing packets of data, the method comprising the steps of:

receiving, at a jitter buffer, a sequence of packets encoded at a first sampling rate by an audio encoder, wherein each packet includes audio data having at least one sample;

determining a clock skew between the audio encoder having the first sampling rate and an audio decoder having a second sampling rate, the clock skew representative of a difference between the first sampling rate and the second sampling rate;

selecting a subset of packets from the sequence of packets based on a magnitude of the clock skew; and

altering each packet of the subset of packets before the packet is provided to the audio decoder to collectively compensate for the clock skew.

2. The method of claim 1, wherein the step of altering includes one or more of:

enlarging the packet when the first sampling rate is greater than the second sampling rate; and

reducing the packet when the first sampling rate is less than the second sampling rate.

3. The method of claim 1, further including the step of providing each packet of the sequence of packets to the audio decoder.

4. The method of claim 1, wherein the step of determining the clock skew includes determining a difference between a desired amount of data and an actual amount of data stored in the jitter buffer, wherein the difference is representative of the clock skew.

5. The method of claim 4, wherein the step of selecting the subset of packets includes the steps of:

determining a number of samples equivalent to a result of an integer division of the difference between the desired amount of data and the actual amount of data by a size of a sample;

selecting every packet of the sequence of packets when the number of samples is at least as great as a number of packets in the sequence of packets; and

selecting a number of packets equivalent to the number of samples from the sequence of packets when the number of samples is less than the number of packets in the sequence of packets.

6. The method of claim 5, wherein the step of selecting a number of packets equivalent to the number of samples includes selecting the number of packets such that the selected packets are substantially evenly distributed among the sequence of packets.

7. The method of claim 4, wherein the step of altering each of the first subset of packets includes the step of altering each packet of the subset by a number of samples, wherein a total number of samples associated with each packet is increased by the number of samples when the actual amount of data is less than the ideal amount of data, and wherein a total number of samples associated with each packet is decreased by the number of samples when the actual amount of data is greater than the ideal amount of data;

and wherein a change in the amount of data represented by the subset of packets after each packet is altered is essentially equivalent to the difference between the desired amount and the actual amount of data in the jitter buffer.

8. The method of claim 1, wherein the step of enlarging the packet includes adding at least one sample to the packet, and wherein the at least one added sample is a copy of another sample of the packet.

9. The method of claim 1, wherein the step of reducing the packet includes removing at least one sample from the packet.

10. The method of claim 1, wherein a sample includes a frame.

11. A method to minimize underflow and overflow of a jitter buffer, the method comprising the steps of:

receiving, at the jitter buffer, a sequence of packets from an audio encoder having a first sampling rate, each packet of the sequence of packets including audio data having at least one sample;

determining a difference between an actual amount of data stored in the jitter buffer and a threshold, wherein the difference represents a clock skew between the first sampling rate and a second sampling rate;

selecting a subset of packets to be altered from the sequence of packets, wherein a number of packets included in the subset is based on the difference;

determining an alter amount based on a relationship between the difference and the number of packets in the subset; and

altering, by the alter amount, each packet of the subset of packets, wherein a total amount of audio data of a packet is decreased when the actual amount of data is greater than the threshold, and wherein the total amount of audio data of a packet is increased when the actual amount of data is less than the threshold.

12. The method of claim 11, wherein the actual amount of data in the jitter buffer is based on a difference between an amount of data provided to the audio decoder and an amount of data received by the jitter buffer.

13. The method of claim 11, wherein the step of selecting the subset of packets to be altered includes determining a number of samples represented by the difference, wherein the number of samples is based on a relationship between a size of a sample and a difference between the actual amount and the threshold, and wherein the number of packets to be altered is equivalent to a minimum value of the number of samples and the number of packets in the sequence of packets.

14. The method of claim 11, wherein increasing the total amount of audio data of the packet includes adding at least one sample to the packet, wherein the at least one added sample is a copy of another sample of the packet.

15. The method of claim 11, wherein decreasing the total amount of audio data of the packet includes removing at least one sample from the packet.

16. The method of claim 11, wherein the alter amount is based on an integer division of the first difference by the number of packets of the subset of packets.

17. The method of claim 11, further including the step of determining a distribution interval, the distribution interval based on a relationship between the number of packets in the sequence of packets and the number of packets in the subset of packets to be altered.

18. The method of claim 17, wherein the distribution interval is based on an integer division of the number of packets in the sequence of packets by the number of packets in the subset of packets.

19. The method of claim 17, wherein the step selecting the subset of packets for altering includes selecting each packet of the sequence of packets located at a position in the sequence of the packets that is an integer multiple of the distribution interval for inclusion into the subset of packets.

20. The method of claim 11, further including the step of providing each packet of the sequence of packets to an audio decoder having the second sampling rate.

21. The method of claim 11, wherein a sample includes a frame.

22. A computer readable medium, said computer readable medium comprising a set of executable instructions adapted to manipulate a processor to:

receive, at a jitter buffer, a sequence of packets encoded at a first sampling rate by an audio encoder, wherein each packet includes audio data having at least one sample;

determine a clock skew between the audio encoder having the first sampling rate and an audio decoder having a second sampling rate, the clock skew representative of a difference between the first sampling rate and the second sampling rate;

select a subset of packets from the sequence of packets based on a magnitude of the clock skew; and

alter each packet of the subset of packets before the packet is provided to the audio decoder to collectively compensate for the clock skew.

23. The computer readable medium of claim 22, wherein the instructions to manipulate the processor to alter each packet include instructions to manipulate the processor to:

enlarge the packet when the first sampling rate is greater than the second sampling rate; and

reduce the packet when the first sampling rate is less than the second sampling rate.

24. A system comprising:

an audio decoder to decode audio data at a first sampling rate;

a jitter buffer coupled to the audio decoder, wherein the jitter buffer is adapted to store a plurality of packets of audio data from an audio encoder, the audio data encoded at a second sampling rate;

a clock skew compensation logic coupled to the audio decoder and the jitter buffer, wherein the clock skew compensation logic is adapted to;

enlarge each packet of a subset of packets by an alter amount when the first sampling rate is greater than the second sampling rate to generate a set of altered packets; and

reduce each packet of a subset of packets by an alter amount when the first sampling rate is less than the second sampling rate to generate a set of altered packets.

25. The system of claim 24, wherein the clock skew compensation logic further is adapted to determine a first difference between an actual amount of audio data and a desired amount of audio data stored in the jitter buffer, where the first difference is representative of a second difference between the first sampling rate and the second sampling rate.

26. The system of claim 25, wherein a third difference between a total amount of audio data of the subset of packets and a total amount of audio data of the set of altered packets is essentially equivalent to the first difference.

27. The system of claim 26, wherein the alter amount is equivalent to a division of the third difference by a number of packets of the subset of packets.

28. The system of claim 25, wherein the clock skew compensation logic further is adapted to determine the first difference for a plurality of alter intervals, wherein each alter interval includes a sequence of packets of the plurality of packets.

29. The system of claim 28, wherein the clock skew compensation logic further is adapted to generate a set of altered packets from a subset of packets from each alter interval of the plurality of alter intervals.

30. The system of claim 24, wherein the clock skew compensation logic further is adapted to select the subset of packets essentially evenly from the plurality of packets.

31. In a system for communicating audio information, a system comprising:

a first data terminal device having an audio encoder, wherein the first data terminal device is adapted to transmit audio data encoded at a first rate over a communications network; and

a second data terminal device coupled to the first data terminal device, the second data terminal having a jitter buffer for receiving incoming audio data over a communications network and an audio decoder for decoding audio data at a second rate, and wherein the second data terminal device is adapted to modify a size of the incoming audio data to compensate for a difference between the first rate and the second rate.

32. The system of claim 31, wherein:

the second data terminal device is adapted to increase the size of the audio data when the first rate is less than the second rate; and

the second data terminal device is adapted to decrease the size of the audio data when the first rate is greater than the second rate.

33. The system of claim 31, wherein:

the first data terminal device further is adapted to transmit the audio data in the form of at least one packet having at least one sample; and

the jitter buffer of the second data terminal device further is adapted to receive at least one packet of audio data.

34. The system of claim 33, wherein the second data terminal device further is adapted to:

remove at least one sample from at least one packet to reduce the size of the audio data when the first rate is greater than the second rate; and

add at least one sample to at least one packet to increase the size of the audio data when the first rate is less than the second rate.

35. The system of claim 33, wherein a sample includes a frame.

36. A data terminal device to process audio data comprising:

a jitter buffer for receiving incoming audio data encoded at a first rate;

a decoder to decode the incoming audio data at a second rate; and

means for modifying a size of the incoming audio data to compensate for a difference between the first rate and the second rate.

37. The data terminal device of claim 36, wherein the means for modifying the size of the incoming audio data include:

means for increasing the size of the incoming audio data when the first rate is less than the second rate; and

means for decreasing the size of the incoming audio data when the first rate is greater than the second rate.

38. The data terminal device of claim 37, wherein the incoming audio data is received by the jitter buffer as at least one packet having at least one sample.

39. The data terminal device of claim 38, wherein the means for modifying a size of the incoming audio data includes:

means for removing at least one sample from at least one packet to reduce the size of the audio data when the first rate is greater than the second rate; and

means for adding at least one sample to at least one packet to increase the size of the audio data when the first rate is less than the second rate.

40. The data terminal device of claim 38, wherein a sample includes a frame.