Method and arrangement in a communication system

• US 7,502,733 B2
• Filed: 07/05/2007
• Issued: 03/10/2009
• Est. Priority Date: 12/28/1999
• Status: Expired due to Fees

• Current Assignee: Google LLC (Alphabet Inc.)
• Original Assignee: Global IP Solutions (GIPS) AB (Alphabet Inc.), Global IP Solutions, Inc. (Alphabet Inc.)
• Inventors: Andrsen, Soren V., Sorqvist, Patrik, Kleijn, Willem B.
• Primary Examiner(s): Armstrong; Angela A
• Application Number:
  US11/822,378
• Publication Number:
  US 20070260462A1
• Time in Patent Office: 614 Days
• Field of Search: 704/201, 704/211, 704/500
• US Class Current: 704/211
• CPC Class Codes: H04L 12/64 Hybrid switching systems

Abstract

The present invention relates to the decoding/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Claims

1. A method for manipulating a sequence of digitized sound signal frames of a sound signal received from a packet switched network, the method including:

storing sound data packets from the packet switched network in a jitter buffer;

decoding sound signal frames from the sound data packets of the jitter buffer;

performing real-time observations on the number of packets in the jitter buffer; and

expanding a decoded first frame, if the real-time observations indicate a state of the jitter buffer close to underflow, by producing a frame portion and adding the produced frame portion to the decoded first frame, wherein the produced frame portion represents a part of the sound signal that is different from the part represented by the decoded first frame.

2. The method of claim 1, wherein

the produced frame portion is produced to be contiguous with the decoded first frame to be expanded;

and

a following decoded second frame is provided to be contiguous with said produced frame portion.

3. The method of claim 1, wherein a time length of the produced frame portion is different from the time length of the decoded first frame to be expanded.

4. The method as claimed in claim 3, wherein the time length of the produced frame portion is chosen such that it provides a smooth transition to said following second frame.

5. The method as claimed in claim 3, wherein the time length of the produced frame portion is chosen based upon a requirement to fulfill a signal fitting criteria with respect to signal characteristics of the sound signal.

6. The method as claimed in claim 3, wherein a resolution of the time length of the produced frame portion is a fraction of the time between two samples of said sound signal, thereby providing an improved signal fitting quality for a smooth transition from the decoded first frame to the produced frame portion.

7. The method as claimed in claim 1, further including matching a true state of a trailing part of the decoded first frame with an entry state of a codebook, the codebook having entries consisting of different states formed by vectors of samples and storing a corresponding signal segment for each entry state, wherein the step of producing a frame portion includes reading out a signal segment from said codebook that corresponds to the entry state that have been matched with said true state.

8. The method as claimed in claim 7, wherein different signal segments of said codebook have different time lengths, thereby enabling continuous transition from the time expanded signal frame to a consecutive signal frame.

9. The method as claimed in claim 7, wherein time delays between states of said entries in said codebook are incremental delays with a resolution of a fraction of a time between two samples.

10. The method as claimed in claim 7, wherein the states of said codebook are scaled in order to improve the matching with said true state, and wherein a trailing part of the signal segment read out from the codebook is scaled to provide a smooth transition to said following second frame.

11. The method as claimed in claim 1, further including:

merging a decoded first frame with a following decoded second frame, if the real-time observations indicate a state of the jitter buffer close to overflow, by merging a heading segment of the first frame with a trailing segment of the second frame using overlap-add, wherein a time-shift of one of the frames is employed for optimizing the matching of the overlapping part of the two segments.

12. The method as claimed in claim 11, wherein said time-shift has a resolution of a fraction of a time between two samples.

13. The method as claimed in claim 11, wherein said heading segment is multiplied with a suitable gain to further optimize the matching with said trailing segment at the overlapping part, after which a smooth transition back to unity gain is performed in order to avoid sound signal discontinuities.

14. A receiver unit for manipulating a sequence of digitized sound signal frames of a sound signal received from a packet switched network, the receiver including

a jitter buffer, storing sound data packets from the packet switched network;

a sound decoder for decoding sound signal frames from the sound data packets of the jitter buffer;

a timing recovery processing unit for performing real-time observations on the number of packets in the jitter buffer; and

the timing recovery processing unit being configured for expanding a decoded first frame, if the real-time observations indicate a state of the jitter buffer close to underflow, by producing a frame portion and adding the produced frame portion to the decoded first frame, wherein the produced frame portion represents a part of the sound signal that is different from the part represented by the decoded first frame.

15. The receiver of claim 14, wherein

the produced frame portion is produced to be contiguous with the decoded first frame to be expanded;

and

a following decoded second frame is provided to be contiguous with said produced frame portion.

16. The receiver of claim 14, wherein a time length of the produced frame portion is different from the time length of the decoded first frame to be expanded.

17. The receiver as claimed in claim 16, wherein the time length of the produced frame portion is configured such that it provides a smooth transition to said following second frame.

18. The receiver as claimed in claim 16, wherein the time length of the produced frame portion is configured based upon a requirement to fulfill a signal fitting criteria with respect to signal characteristics of the sound signal.

19. The receiver as claimed in claim 16, wherein a resolution of the time length of the produced frame portion is a fraction of the time between two samples of said sound signal, thereby providing an improved signal fitting quality for a smooth transition from the decoded first frame to the produced frame portion.

20. The receiver as claimed in claim 14, further including a codebook, wherein the timing recovery processing unit is configured for matching a true state of a trailing part of the decoded first frame with an entry state of the codebook, the codebook having entries consisting of different states formed by vectors of samples and storing a corresponding signal segment for each entry state, wherein the producing of a frame portion includes reading out a signal segment from said codebook that corresponds to the entry state that have been matched with said true state.

21. The receiver as claimed in claim 20, wherein different signal segments of said codebook have different time lengths, thereby enabling continuous transition from the time expanded signal frame to a consecutive signal frame.

22. The receiver as claimed in claim 20, wherein time delays between states of said entries in said codebook are incremental delays with a resolution of a fraction of a time between two samples.

23. The receiver as claimed in claim 20, wherein the states of said codebook are scaled in order to improve the matching with said true state, and wherein a trailing part of the signal segment read out from the codebook is scaled to provide a smooth transition to said following second frame.

24. The receiver as claimed in claim 14, wherein the timing recovery processing unit is further configured for

merging a decoded first frame with a following decoded second frame, if the real-time observations indicate a state of the jitter buffer close to overflow, by merging a heading segment of the first frame with a trailing segment of the second frame using overlap-add, wherein a time-shift of one of the frames is employed for optimizing the matching of the overlapping part of the two segments.

25. The receiver as claimed in claim 24, wherein said time-shift has a resolution of a fraction of a time between two samples.

26. The receiver as claimed in claim 24, wherein said heading segment is multiplied with a suitable gain to further optimize the matching with said trailing segment at the overlapping part, after which a smooth transition back to unity gain is performed in order to avoid sound signal discontinuities.