Method and system for adjusting digital audio playback sampling rate

• US 7,650,285 B2
• Filed: 06/25/2004
• Issued: 01/19/2010
• Est. Priority Date: 06/25/2004
• Status: Expired due to Fees

• Current Assignee: Numerex Corporation (Sierra Wireless Incorporated)
• Original Assignee: Numerex Corporation (Sierra Wireless Incorporated)
• Inventors: Magliaro, Max, Panulla, Gary
• Primary Examiner(s): Dorvil; Richemond
• Assistant Examiner(s): Jackson; Jakieda R
• Application Number:
  US10/877,354
• Publication Number:
  US 20060009983A1
• Time in Patent Office: 2,034 Days
• Field of Search: 704/500
• US Class Current: 704/500
• 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

In a packet communication network, to compensate for rate mismatches between transmitting and receiving devices, an apparatus and method provides for adjusting the playback sampling rate and for monitoring the buffer. The receiving device will monitor its buffer; relevant buffer data can comprise whether the buffer is approaching capacity or approaching depletion and the speed in which the buffer is approaching capacity of approaching depletion. The receiving device will then trigger an adjustment to the playback sampling rate to attune the rates of the transmitting and receiving devices or to compensate for jitters from any number of network complications. The receiving device may also store the buffer data for later action, for example, to formulate specific adjustment procedures or to compile specific conference profiles. The present invention may function alone or function in conjunction with other known methods in the art.

Claims

1. An apparatus for facilitating real-time audio data communication 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, for temporarily storing the data packets;

a digital to analog converter, coupled to the buffer, for converting digital audio data in the data packets to an analog signal;

a clocking mechanism, coupled to the digital to analog converter, for providing variable frequencies to the digital to analog converter; and

a buffer monitor for monitoring the buffer'"'"'s activity during the real-time audio data communication, wherein the buffer monitor adjusts the playback sampling rate when the buffer approaches a capacity trigger value or a depletion trigger value;

and wherein the buffer monitor has a buffer capacity such that;

if an average of the buffer capacity is greater than 90%, the playback sampling rate is increased by 4 Hz;

if the avenge of the buffer capacity is greater than 80%, the playback sampling rate is increased by 2 Hz;

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

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

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

3. The apparatus of claim 1, further comprising an analog to digital converter.

4. The apparatus of claim 1, further comprising an encoder.

5. The apparatus of claim 1, wherein the buffer monitor is further operable for calculating the average number of data packets in the buffer.

6. The apparatus of claim 5, further comprising a digital signal processor.

7. The apparatus of claim 5, further comprising read-only memory or Flash memory for storing data.

8. The apparatus of claim 5, further comprising an interface to a Peripheral Component Interconnect slot of a personal computer.

9. The apparatus of claim 5, further comprising in interface to an Industry Standard Architecture bus of a personal computer.

10. The apparatus of claim 5, wherein the buffer monitor is a software application stored in the Read Only Memory in a personal computer.

11. The apparatus of claim 10, wherein further adjustments to the playback sampling rate range between 0.0 and 8.0 Hz.

12. The apparatus 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 after different periods; and

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

13. The apparatus of claim 1, wherein the playback sampling rate is approximately 8000 Hz.

14. The apparatus of claim 1, wherein the data packet network comprises a TCP/IP network.

15. The apparatus of claim 1, wherein the data interface further comprises a socket interface for establishing socket connections.

16. The apparatus of claim 1, wherein the digital to analog converter resides on a sound card.

17. A system for facilitating real-time data networking of audio data over a network comprising:

a transmitter comprising;

an analog to digital converter for converting an analog audio signal to digital data;

a clocking mechanism for providing a frequency to the analog to digital converter that establishes the analog to digital converter'"'"'s sampling rate; and

an interface coupled to the analog to digital converter for transmitting the digital data over the packet network;

a receiver comprising;

an interface for receiving digital data transmitted over the packet network;

a digital to analog converter for converting the digital data to an analog signal;

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

a buffer that temporarily stores the digital data; and

a buffer monitor for;

querying the buffer for determining the buffer'"'"'s capacity; and

triggering an adjustment in the playback sampling rate such that;

if an average of the buffer capacity is greater tan 90%, the playback sampling rate is increased by 4 Hz;

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

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

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

18. The system of claim 17, wherein the digital data comprise frames, and wherein the sender and the receiver further comprise encoders.

19. The system of claim 17, wherein the transmitter and the receiver are personal computers.

20. The system of claim 19, wherein the buffer monitor is a computer application that resides in the personal computer'"'"'s hard drive.

21. The system of claim 17, wherein the buffer'"'"'s capacity comprises the avenge amount of digital data in the buffer.

22. The system of claim 17, wherein the receiver and transmitter'"'"'s sampling rates are approximately 8000 Hz.

23. The system of claim 22, wherein further adjustments to the playback sampling rate range between 0.0 and 10.0 Hz.

24. The system of claim 22, wherein the transmitter and the receiver communicate via sockets.

25. A method of adjusting playback sampling rate to facilitate real-time audio communication over a packet network comprising the steps of:

receiving packets at a network interface;

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

monitoring the buffer'"'"'s capacity;

forwarding packets from the buffer to a digital to analog converter for conversion to an analog signal for playback at a sampling rate;

determining whether to adjust the playback sampling rate based on the buffer'"'"'s capacity such that;

if an average of the buffer capacity is greater than 90%, the playback sampling rate is increased by 4 Hz;

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

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

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

26. The method of claim 25, wherein the step of monitoring the buffer'"'"'s capacity further comprises querying the buffer upon receiving each data packet.

27. The method of claim 25, wherein the step of monitoring the buffer'"'"'s capacity further comprises querying the buffer upon receiving each data packet to determine the number of data packets stored in the buffer.

28. The method of claim 25, wherein the step of monitoring the buffer'"'"'s capacity comprises the steps of:

upon receiving each data packet, querying the buffer to determine the number of data packets stored in the buffer;

updating a variable that sums the query results; and

updating a variable that sums the number of packets received.

29. The method of claim 25, wherein the step of monitoring the buffer'"'"'s capacity further comprises reporting data to a buffer monitor.

31. The method of claim 25, wherein the step of determining whether to adjust the playback sampling rate further comprises the step of determining an average number of packets in the buffer.

30. The method of 25, 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.

32. A method for adjusting a playback sampling rate of a receiver to compensate for variations in buffering, sampling, and clock accuracy during real-time audio communication sessions over a packet network comprising the steps of:

receiving packets over a packet network at a network interface;

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

querying the buffer to determine the number of packets that are stored in the buffer;

summing the number of packets that are stored in the buffer;

summing the total number of packets received;

comparing the number of packets stored in the buffer to a capacity of the buffer; and

determining whether to adjust the playback sampling rate according to the query results such that;

if the number of packets in the buffer is greater than 90% of the capacity of the buffer, the playback sampling rate is increased by 4 Hz;

if the number of packets in the buffer is greater than 80% of the capacity, the playback sampling rate is increased by 2 Hz;

if the number of packets in the buffer is less than 10%, of the capacity, the playback sampling rate is decreased by 4 Hz; and

if the number of packets in the buffer is less than 20% of the capacity, the playback sampling rate is decreased by 2 Hz.

33. The method of claim 32, wherein if it is determined that the playback sampling rate is to be adjusted, playing an audio signal by converting the received packets at the adjusted playback sampling rate to an audio stream.

34. The method of claim 32, wherein the packets comprise frames, and the method further comprises the step of decoding the frames.

35. The method of claim 32, wherein the packet network comprises a TCP/IP network.

36. A method of transmitting audio data to a receiver comprising the step of:

transmitting audio data to the receiver in data packets wherein the receiver is operable for;

receiving the data packets at a network interface;

storing the data packets in a buffer;

determining an average number of data packets in the buffer;

comparing the average number of data packets to a capacity of the buffer; and

determining a receiver'"'"'s playback sampling rate for the audio data based on the average number of data packets in the buffer such that;

if the average number of packets in the buffer is greater than 90% of the capacity of the buffer, the playback sampling rate is increased by 4 Hz;

if the average number of packets in the buffer is greater than 80% of the capacity, the playback sampling rate is increased by 2 Hz;

if the average number of packets in the buffer is less than 10%, of the capacity, the playback sampling rate is decreased by 4 Hz; and

if the average number of packets in the buffer is less than 20% of the capacity, the playback sampling rate is decreased by 2 Hz.

37. The method of claim 36, further comprising the steps of:

converting an analog signal to digital data at a transmitter'"'"'s sampling rate; and

placing the digital data in packets in accordance wit a packet network'"'"'s protocols.