Method And System For Improving Real-Time Data Communications

• US 20100091769A1
• Filed: 12/17/2009
• Published: 04/15/2010
• Est. Priority Date: 06/25/2004
• Status: Active Grant

• Current Assignee: Numerex Corporation (Sierra Wireless Incorporated)
• Original Assignee: Numerex Corporation (Sierra Wireless Incorporated)
• Inventors: Magliaro, Max, Panulla, Gary
• Granted Patent: US 8,112,285 B2
• Time in Patent Office: Days
• US Class Current: 370/352
• CPC Class Codes: G10L 19/167 Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• CPC Class Codes: G10L 19/24 Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Abstract

A system and method for improving real-time data communications by accounting for sampling rate mismatches between a transmitter and a receiver. Based on an analysis of the average number of packets received at a receiver over a period of time, a buffer monitor cooperating with the receiver can trigger an adjustment to the playback sampling rate to account for mismatches in the sampling rates of the transmitter and receiver. The buffer monitor may adjust the playback sampling rate more dramatically if the average is dangerously high or low, adjust the playback sampling rate less dramatically if the average is near satisfactory conditions, and not adjust the playback sampling rate if the average falls is satisfactory.

Claims

1. A system for adjusting a playback sampling rate for real-time data communications over a data packet network, comprising:

a data interface for receiving data packets from the data packet network;

a buffer coupled to the data interface and configured to temporarily store the data packets;

a digital to analog converter coupled to the buffer and configured to convert the data packets to an analog signal;

a clocking mechanism coupled to the digital to analog converter and configured to provide the digital to analog converter with variable frequencies;

a buffer monitor for monitoring the buffer'"'"'s activity during the real-time audio data communications, wherein the buffer monitor is configured to adjust the playback sampling rate; and

a timer for preventing the adjustment of the playback sampling rate by the buffer monitor until after the expiration of a pre-determined period of time.

2. The system of claim 1, wherein the data packets comprise frames.

3. The system of claim 1, wherein the data packets comprise audio transmitted during a Voice over Internet Protocol communication.

4. The system of claim 1, wherein the buffer monitor is further configured to calculate the average number of data packets stored in the buffer over the pre-determined period of time.

5. The system of claim 1, wherein the buffer monitor is further operable for:

calculating a plurality of averages for the number of data packets in the buffer; and

determining an adjustment to the playback sampling rate based on the plurality of averages.

6. The system of claim 5, wherein the playback sampling rate is increased if the plurality of averages is greater than 80% of a capacity of the buffer and the playback sampling rate is decreased if the plurality of averages is less than 20% of the capacity of the buffer.

7. The system of claim 4, wherein the playback sampling rate is adjusted by 8 Hz when the average is high or low, adjusted by 2 Hz if the average is near satisfactory conditions, and is not adjusted when the average falls in a satisfactory zone.

8. The system of claim 1, wherein an adjustment to the playback sampling rate comprises one of 2.0, 4.0, 6.0, and 8.0 Hz.

9. The system of claim 1, wherein an adjustment to the playback sampling rate is not performed until after ten seconds have elapsed since the arrival of the first data packet.

10. The system of claim 1, wherein the buffer monitor is only allowed to adjust the playback sampling rate after twenty seconds have elapsed since the last adjustment of the playback sampling rate.

11. The system of claim 4, wherein determining an adjustment to the playback sampling rate comprises the following:

when the average number of data packets in the buffer is greater than 4.5, the playback sampling rate is increased by 4 Hz;

when the average number of data packets in the buffer is greater than 4.0 but less than or equal to 4.5, the playback sampling rate is increased by 2 Hz;

when the average number of data packets in the buffer is between or equal to 4.0 and 1.5, the playback sampling rate is not adjusted;

when the average number of data packets in the buffer is less than 1.5 but greater than or equal to 0.5, the playback sampling rate is decreased by 2 Hz; and

when the average amount of data packets in the buffer is less than 0.5, the playback sampling rate is decreased by 4 Hz.

12. A system for accounting for variances in sampling rates in a transmitter and a receiver communicating over a packet network, comprising:

an interface at the receiver for receiving and decoding data packets transmitted over the packet network;

a digital to analog converter at the receiver configured to convert the data packets to an analog signal;

a clocking mechanism at the receiver for providing a frequency to the digital to analog converter that establishes the receiver'"'"'s playback sampling rate, wherein the clocking mechanism is configured to provide varying frequencies to the digital to analog converter;

a buffer at the receiver that temporarily stores the data packets; and

a buffer monitor at the receiver configured to;

determine the average number of data packets stored in the buffer over a given time period; and

based on the determination, trigger an adjustment in the playback sampling rate for the receiver to account for the variances between the receiver'"'"'s sampling rate and the transmitter'"'"'s sampling rate.

13. The system of claim 12, wherein adjustments to the playback sampling rate are not performed until after ten seconds have elapsed since the arrival of the first data packet.

14. The system of claim 12, wherein adjustments to the playback sampling rate are made as follows:

when the average number of data packets in the buffer over the time period is greater than 4.5, the playback sampling rate is increased by 4 Hz;

when the average number of data packets in the buffer over the time period is greater than 4.0 but less than or equal to 4.5, the playback sampling rate is increased by 2 Hz;

when the average number of data packets in the buffer over the time period is between or equal to 4.0 and 1.5, the playback sampling rate is not adjusted;

when the average number of data packets in the buffer over the time period is less than 1.5 but greater than or equal to 0.5, the playback sampling rate is decreased by 2 Hz; and

when the average number of data packets in the buffer over the time period is less than 0.5, the playback sampling rate is decreased by 4 Hz.

15. A method for adjusting a playback sampling rate, comprising the steps of:

receiving packets over the packet network at a network interface;

forwarding the packets from the network interface to a buffer for temporary storage;

querying the buffer with a buffer monitor to determine the average number of packets stored in the buffer over a specified time interval;

determining whether the capacity of the buffer is approaching capacity or depletion based on the average number of packets stored in the buffer; and

adjusting the playback sampling rate for the receiver based on the determination.

16. The method of claim 15, further comprising the step of:

if the buffer approaches capacity, increase the playback sampling by between approximately 2 Hz and 4 Hz.

17. The method of claim 15, further comprising the step of:

if the buffer approaches depletion, decrease the playback sampling rate by between approximately 2 Hz and 4 Hz.

18. The method of claim 15, further comprising the steps of:

if the buffer capacity is on average greater than 90%, increase the playback sampling rate by 4 Hz;

if the buffer capacity is on average greater than 80%, increase the playback sampling rate by 2 Hz;

if the buffer capacity is on average less than 10%, decrease the playback sampling rate by 4 Hz; and

if the buffer capacity is on average less than 20%, decrease the playback sampling rate by 2 Hz.

20. The method of claim 15, wherein the playback sampling rate is only adjusted after twenty seconds have elapsed since the last adjustment of the playback sampling rate.

19. The method of 15, further comprising the step of determining the amount to increase or decrease the playback sampling rate according to the duration of time in which the buffer took to approach capacity or to approach depletion.