Method for tuning voice playback ratio to optimize call quality

• US 20030115320A1
• Filed: 12/19/2001
• Published: 06/19/2003
• Est. Priority Date: 12/19/2001
• Status: Abandoned Application

• Current Assignee: Motorola, Inc. (Motorola Solutions, Inc.)
• Original Assignee: Motorola, Inc. (Motorola Solutions, Inc.)
• Inventors: Chen, Edward, Yarroll, Lamonte H.P.
• Application Number:
  US10/024,797
• Publication Number:
  US 20030115320A1
• Time in Patent Office: Days
• US Class Current: 709/224
• CPC Class Codes: G10L 25/69 for evaluating synthetic or decoded voice signals
• 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

Abstract

An apparatus and method for tuning voice playback ratio (pbr) to optimize call quality in a packet voice communications system, while taking into account network conditions. The pbr is the ratio of resampling rate to the original sampling rate. The invention also optimizes jitter buffer length (jb₀) for call quality. Between bursts of speech, the preferred embodiment of the invention optimizes call quality by varying the initial jb₀ and the pbr to achieve the best R-factor (R). R is an estimate of customer satisfaction with the quality of a voice call in real time. During bursts of speech, the value of jb₀ is fixed at the beginning of the BOS and the pbr is varied to achieve the best R. The method can be implemented during a burst of speech and between bursts of speech whenever the network conditions change.

Claims

1. A method for optimizing customer experience of a real-time system comprising:

collecting statistics from a network;

using the statistics to choose a plurality of parameters;

using the plurality of parameters to manipulate playback properties of the real-time system to optimize the customer experience as measured on a physiological.

2. The method of claim 1 wherein the step of collecting statistics from a network comprises measuring network conditions delay, jitter and loss.

3. The method of claim 2 wherein the step of using the statistics to choose a plurality of parameters comprises for the measured delay, jitter and loss, determining a jitter buffer length and a playback ratio that yield a best R-factor, wherein the R-factor is determined by the equation R=R₀−

Ie_c−

Ie_pbr−

Ie_DD.

4. A method of optimizing jitter buffer length and playback ratio to improve call quality comprising the steps of:

measuring network conditions delay, jitter and loss; and

for the measured delay, jitter and loss, determining a jitter buffer length and a playback ratio that yield a best R-factor, wherein the R-factor is determined by the equation R=R₀−

Ie_c−

Ie_loss−

Ie_pbr−

Ie_DD.

5. The method of claim 4 wherein the step of determining a jitter buffer length and a playback ratio that yield the best R-factor comprises the steps of:

a) setting the jitter buffer length and the playback ratio to an initial value;

b) determining R₀;

c) determining Ie_c;

d) determining Ie_loss;

e) determining Ie_phr;

f) determining Ie_DD;

g) calculating R=R₀−

Ie_c−

Ie_loss−

Ie_pbr−

Ie_DD;

h) determining whether an optimum value of R has been achieved; and

i) when the optimum value of R has not been achieved, changing the value of jitter buffer length and/or playback ratio and repeating steps b through h.

6. The method of claim 5 wherein the step of determining Ie_loss comprises:

determining an initial playback time;

determining an initial jitter buffer overflow;

using the initial jitter buffer overflow to determine an initial jitter buffer loss;

determining a gain in jitter buffer length;

using the gain in jitter buffer length to determine a final playback time;

determining a final jitter buffer overflow;

using the final jitter buffer overflow to determine a final jitter buffer loss;

determining an average jitter buffer loss from the initial jitter buffer loss and the final jitter buffer loss; and

using the average jitter buffer loss to determine Ie_loss.

7. The method of claim 6 wherein the step of determining an initial playback time comprises solving the equation initial pbt=jb₀+delay.

8. The method of claim 6 wherein the step of using the initial jitter buffer overflow to determine an initial jitter buffer loss comprises solving the equation initial jb_loss=1−

[(1−

loss)×

(1−

initial jitter buffer overflow)].

9. The method of claim 6 wherein the step of determining a gain in jitter buffer length comprises solving the equation gain in jitter buffer length=(1−

pbr)×

BOS.

10. The method of claim 6 wherein the step of using the gain in jitter buffer length to determine a final playback time comprises solving the equation final pbt=jb₀+delay+gain in jitter buffer length.

11. The method of claim 6 wherein the step of using the final jitter buffer overflow to determine a final jitter buffer loss comprises solving the equation final jb_loss=1−

[(1−

loss)×

(1−

final jitter buffer overflow)].

12. An apparatus for optimizing customer experience of a real-time system comprising:

a device for collecting statistics from the network;

a control apparatus operatively coupled to the device for manipulating playback properties of the real-time system; and

an optimizer operatively coupled to the device for using the statistics to choose a plurality of parameters for the control apparatus, wherein the plurality of parameters are chosen to optimize the customer experience as measured on a physiological scale.

13. An apparatus for optimizing jitter buffer length to improve call quality comprising:

a jitter buffer;

a voice decoder operatively coupled to the jitter buffer for controlling a rate at which voice date is removed from the jitter buffer;

a voice resampler operatively coupled to the voice decoder for controlling a number of bits removed from the voice decoder; and

a playback optimizer operatively coupled to the jitter buffer and the voice resampler for receiving statistics on a communication link from the jitter buffer, for using the statistics to determine a jitter buffer length and playback ratio that yield an optimum score on a physiological scale and for sending the jitter buffer length and playback ratio to the voice resampler to improve call quality.