Methods for changing the size of a jitter buffer and for time alignment, communications system, receiving end, and transcoder

• US 20040120309A1
• Filed: 12/29/2003
• Published: 06/24/2004
• Est. Priority Date: 04/24/2001
• Status: Abandoned Application

• Current Assignee: Nokia Corporation
• Original Assignee: Nokia Corporation
• Inventors: Kirla, Olli, Kurittu, Antti
• Application Number:
  US10/475,779
• Publication Number:
  US 20040120309A1
• Time in Patent Office: Days
• US Class Current: 370/352
• CPC Class Codes: G10L 19/00 Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis in musical instruments G10H
• CPC Class Codes: G10L 21/04 Time compression or expansion
• CPC Class Codes: H04J 3/0632 Synchronisation of packets and cells, e.g. transmission of voice via a packet network, circuit emulation service [CES] queuing arrangements in packet switching elements H04L49/90; synchronising systems for the synchronous transmission of a pulse code modulated video signal with one or more other pulse code modulated signals H04N7/56
• CPC Class Codes: H04W 88/181 Transcoding devices; Rate adaptation devices

Abstract

The invention relates to a method for changing the size of a jitter buffer, which jitter buffer is employed at a receiving end in a communications system including a packet network for buffering received packets containing audio data in order to enable a compensation of varying delays of said received packets. In order to enable a faster change of the jitter buffer size it is proposed that in case it is determined that the current jitter buffer size has to be changed, the jitter buffer size is expanded by generating additional data based on the received data or decreased by compacting the received data. A proposed communications system, receiving end and processing unit include corresponding means. The invention equally relates to a method for time alignment in a radio communications system based on existing speech data. A further proposed communications system, transceiver unit and processing unit include the corresponding means.

Claims

1. Method for changing the size of a jitter buffer, which jitter buffer is employed at a receiving end in a communications system including a packet network for buffering received packets containing audio data in order to enable a compensation of varying delays of said received packets, the method comprising:

determining whether a current jitter buffer size should be increased or decreased by evaluating current overall delay and jitter in received packets; and

in case it was determined that the current jitter buffer size is to be increased, increasing the jitter buffer size and compensating the resulting empty jitter buffer space by generating additional data based on audio data contained in received packets.

2. Method according to claim 1, wherein in case it was determined that the current jitter buffer size is to be decreased, decreasing the jitter buffer size by condensing at least part of the audio data currently present in the jitter buffer.

3. Method according to one of the preceding claims, wherein the audio data buffered in the jitter buffer is decoded audio data.

4. Method according to one of the preceding claims, wherein said additional data is generated by treating said empty jitter buffer space as lost frames and by compensating said empty jitter buffer space by a bad frame handling.

5. Method according to one of the preceding claims, wherein additional data is generated by replicating a waveform that starts a whole multitude of pitch period lengths before a point of time at which said additional data is to be added.

6. Method according to one of the preceding claims, wherein said additional data is generated to overlap part of received audio data in the jitter buffer.

7. Method according to one of the preceding claims, wherein at least part of said additional data is attenuated in case more than a predetermined amount of additional data is generated.

8. Method according to one of the preceding claims, wherein data is condensed by overlapping the audio data of a selected first packet (32) and the audio data of a selected second packet (33) and by discarding the packets in between said first and said second packet (32, 33).

9. Method according to claim 8, wherein audio data is overlapped by multiplying the audio data of a selected first packet (32) with a downramp function (36) and the audio data of a selected second packet (33) with an upramp function (37), by adding the multiplied data of said first and said second packet (32, 33), and by discarding the packets in between said first and said second packet (32, 33).

10. Method according to one of claims 1 to 3, wherein for changing the jitter buffer size, additional data is generated and/or audio data in said jitter buffer is condensed by a time domain time scaling performed on at least part of the audio data currently buffered in the jitter buffer for expanding or compressing said at least part of the audio data respectively.

11. Method according to claim 10, wherein the time domain time scaling comprises

a) selecting received audio data currently buffered in the jitter buffer, which audio data is to be used for time scaling;

b) applying a time domain time scaling method on the selected audio data for obtaining time scaled audio data; and

c) substituting in the jitter buffer the time scaled audio data for at least part of the selected received audio data.

12. Method according to claim 11, wherein the time scaling applied to selected received audio data is extended for a predetermined length surpassing the length to be used for substituting received audio data in the jitter buffer, which extension is overlapped with the following received audio data.

13. Method according to one of claims 10 to 12, wherein the time domain time scaling is based on a waveform similarity overlap add (WSOLA) method.

14. Method according to one of claims 1 to 3, wherein for changing the jitter buffer size additional data is generated and/or audio data in said jitter buffer is condensed by a frequency domain time scaling performed on at least part of the audio data currently buffered in the jitter buffer for expanding or compressing said at least part of the audio data respectively.

15. Method according to claim 14, wherein the frequency domain time scaling comprises:

Fourier transforming overlapping windowed parts (71-74) of at least part of the received audio data currently contained in the jitter buffer;

time scale modifying the Fourier transformed audio data of each window according to a required amount of increase or decrease of the jitter buffer size; and

inverse Fourier transforming the Fourier transformed and time scale modified audio data of each window.

16. Method according to claim 14 or 15, wherein the frequency domain time scaling is phase vocoder based.

17. Method according to one of claims 1 to 2, wherein the audio data buffered in the jitter buffer is parametric coded audio data.

18. Method according to claim 17, wherein for increasing the jitter buffer size additional data is generated by using a bad frame handler.

19. Method according to claim 17 or 18, wherein for decreasing the jitter buffer size the audio data in the jitter buffer is condensed by discarding at least one frame in the jitter buffer.

20. Method according to claim 19, wherein gain parameters and Linear Predictive Coding (LPC) coefficients of the frames surrounding the at least one discarded frame are modified to smoothly combine the frames surrounding the at least one discarded frame.

21. Method according to claim 17, wherein for increasing the jitter buffer size, additional audio data is interpolated from adjacent data.

22. Method according to claim 17, wherein for decreasing the jitter buffer size, selected adjacent or spaced apart audio data is interpolated into reduced audio data.

23. Communications system comprising a packet network and at least one possible receiving end, the receiving end including:

a jitter buffer for buffering received packets containing audio data;

processing means for compensating varying delays of received packets buffered in said jitter buffer;

processing means for determining whether the size of said jitter buffer should be increased or decreased-based on the current overall delay and the current variation of delay between the different packets; and

processing means for changing the current size of said jitter buffer according to one of the methods 1 to 22.

24. Communications system according to claim 23, wherein the receiving end further includes a bad frame handler for compensating packets lost during transmission to said receiving end, which bad frame handler is moreover employed for generating additional data for an increase of the jitter buffer size by the processing means for changing the current size of said jitter buffer.

25. Receiving end for a communications system including a packet network, which receiving end comprises:

a jitter buffer for buffering received packets containing audio data;

processing means for compensating varying delays of received packets buffered in said jitter buffer;

processing means for determining whether the size of said jitter buffer should be increased or decreased based on the current overall delay and the current variation of delay between the different packets; and

processing means for changing the current size of said jitter buffer according to one of the methods 1 to 22.

26. Receiving end according to claim 25, further including a bad frame handler for compensating packets lost during transmission to said receiving end, which bad frame handler is moreover employed for generating additional data for an increase of the jitter buffer size by the processing means for changing the current size of said jitter buffer.

27. Processing means for a receiving end of a communications system including a packet network, which processing means are designed for changing the current size of a jitter buffer according to one of the methods 1 to 22.

28. Method for carrying out a time alignment in a network transcoder (145) of a radio communications system, which time alignment is used for decreasing a buffering delay in downlink direction, said buffering delay resulting from buffering downlink speech data encoded by said transcoder (145) before transmitting said speech data over a radio interface of said radio communications system in order to compensate for a phase shift in downlink framing of speech data at said transcoder (145) and at said radio interface, the method comprising:

determining whether a time alignment has to be carried out; and

in case it was determined that a time alignment has to be carried out, expanding or compacting selected speech data with a time scaling method for achieving the required time alignment.

29. Method according to claim 28, wherein the time scaling method is a time domain time scaling method comprising

a) selecting original speech data which are to be used for time scaling;

b) applying a time domain time scaling method on the selected speech data for obtaining time scaled speech data; and

c) substituting the time scaled speech data for at least part of the selected speech data.

30. Method according to claim 28, wherein the time scaling applied to selected speech data is extended for a predetermined length surpassing the length to be used for substituting original speech data, which extension is overlapped with the following original speech data.

31. Method according to one of claims 28 to 30, wherein the time domain time scaling is based on a waveform similarity overlap add (WSOLA) method.

32. Method according to claim 28, wherein the time scaling is a frequency domain time scaling.

33. Method according to claim 32, wherein the frequency domain time scaling comprises:

Fourier transforming overlapping windowed parts of selected speech data;

time scale modifying the Fourier transformed speech data of each window according to the required time alignment; and

inverse Fourier transforming the Fourier transformed and time scale modified speech data of each window.

34. Method according to claim 32 or 33, wherein the frequency domain time scaling is phase vocoder based.

35. Method for carrying out a time alignment in a network transcoder of a radio communications system, which time alignment is used for decreasing a buffering delay in downlink direction, said buffering delay resulting from buffering downlink speech data encoded by said transcoder before transmitting said speech data over a radio interface of said radio communications system in order to compensate for a phase shift in a downlink framing of said speech data in said transcoder and at said radio interface, the method comprising:

determining whether a time alignment has to be carried out; and

in case it was determined that a time alignment has to be carried out, expanding selected speech data for achieving the required time alignment by inserting an empty space within said selected speech data and by compensating said empty space by a bad frame handling.

36. Method according to claim 35, wherein additional data is generated by replicating a waveform that starts a whole multitude of pitch period lengths before a point of time at which said additional data is to be added.

37. Method according to one of the claims 35 or 36, wherein said additional data is generated to overlap part of received audio data in the jitter buffer.

38. Method according to one of claims 35 to 37, wherein at least part of said additional data is attenuated in case more than a predetermined amount of additional data is generated.

39. Method for carrying out a time alignment in a network transcoder of a radio communications system, which time alignment is used for decreasing a buffering delay in downlink direction, said buffering delay resulting from buffering downlink speech data encoded by said transcoder before transmitting said speech data over a radio interface of said radio communications system in order to compensate for a phase shift in a downlink framing of said speech data in said transcoder and at said radio interface, the method comprising:

determining whether a time alignment has to be carried out; and

in case it was determined that a time alignment has to be carried out, condensing speech data for achieving the required time alignment by overlapping a selected first portion of speech data and a selected second portion of speech data and by discarding the speech data in between said first and said second selected portion of the speech data.

40. Method according to claim 39, wherein speech data is overlapped by multiplying the speech data of the selected first portion of speech data with a downramp function and the speech data of the selected second portion of speech data with an upramp function, by adding the multiplied data of said first and said second selected portion, and by discarding the speech data in between said first and said second portion.

41. Method for carrying out a time alignment in a network transcoder of a radio communications system, which time alignment is used for decreasing a buffering delay in downlink direction, said buffering delay resulting from buffering downlink speech data encoded by said transcoder before transmitting said speech data over a radio interface of said radio communications system in order to compensate for a phase shift in a downlink framing of said speech data in said transcoder and at said radio interface, the method comprising:

determining whether a time alignment has to be carried out; and

in case it was determined that a time alignment has to be carried out, condensing speech data for achieving the required time alignment by discarding at least one frame of speech data, wherein gain parameters and Linear Predictive Coding (LPC) coefficients of frames of speech data surrounding the at least one discarded frame are modified to smoothly combine the frames surrounding the at least one discarded frame.

42. Method for carrying out a time alignment in a network transcoder of a radio communications system, which time alignment is used for decreasing a buffering delay in downlink direction, said buffering delay resulting from buffering downlink speech data encoded by said transcoder before transmitting said speech data over a radio interface of said radio communications system in order to compensate for a phase shift in a downlink framing of said speech data in said transcoder and at said radio interface, the method comprising:

determining whether a time alignment has to be carried out; and

in case it was determined that a time alignment has to be carried out, expanding speech data for achieving the required time alignment by interpolating additional audio data from selected speech data.

43. Method for carrying out a time alignment in a network transcoder of a radio communications system, which time alignment is used for decreasing a buffering delay in downlink direction, said buffering delay resulting from buffering downlink speech data encoded by said transcoder before transmitting said speech data over a radio interface of said radio communications system in order to compensate for a phase shift in a downlink framing of said speech data in said transcoder and at said radio interface, the method comprising:

determining whether a time alignment has to be carried out; and

in case it was determined that a time alignment has to be carried out, expanding speech data for achieving the required time alignment by interpolating selected adjacent or spaced apart speech data into reduced speech data.

44. Radio communications system comprising

at least one radio interface for transmitting encoded speech data in a downlink direction;

at least one network transcoder (145), which network transcoder (145) includes at least one encoder (146) for encoding speech data to be used for a downlink transmission via said radio interface, and which network transcoder (145) further includes processing means (147) for carrying out a time alignment on encoded speech samples according to one of methods 28 to 43; and

buffering means (143) arranged between said radio interface and said network transcoder (145) for buffering downlink speech data encoded by said transcoder (145) before transmitting said encoded speech data via said radio interface in order to compensate for a phase shift in a downlink framing of said speech data by said transcoder (145) and by said radio interface; and

processing means (143) for determining whether and to which extend the speech samples encoded by said encoder (146) have to be time aligned before transmission in order to minimize a buffering delay for encoded speech data resulting from a buffering by said buffering means (143).

45. Network transcoder (145) for a radio communications system comprising:

at least one encoder (146) for encoding speech data to be used for a downlink transmission via a radio interface of said radio communications system; and

processing means (147) for carrying out a time alignment according to one of the methods 28 to 43.