Low-latency audio interface for packet telephony
First Claim
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1. In a packet telephony communication system, a method for handing audio data for output to a packet network, comprising the steps of:
- a. selecting an interim data rate for processing audio information, wherein said interim rate is substantially higher than an output data rate to reduce latency in a buffer;
b. converting audio information to audio data having a data rate equal to the selected interim data rate;
c. buffering in said buffer the converted audio data stream; and
d. downsampling the data rate of the buffered audio data stream to a rate equal to the output data rate, such flat performing said selecting, converting, buffering and downsampling steps results in reducing latency.
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Abstract
In a method for reducing latency in packet telephony caused by buffering at the conversion stage between analog audio signals and digital audio data, analog audio is sampled at a rate far greater than necessary for telephony. The increased sampling rate allows the audio data to pass much more rapidly through the data conversion buffer. After passing through the buffer, the data is downsampled to a rate normally used for telephony. To handle audio data for speaker output, the data is upsampled to a rate far in excess of the rate necessary for processing telephony-grade voice signals. The increased sampling rate allows the audio data to pass much more rapidly through the data conversion buffer. After passing through the buffer, the data is converted into an analog audio signal for sending to the speaker. In this way, latency due to the buffering that accompanies the process of converting audio signals to digital data, or vice versa, is minimized.
36 Citations
48 Claims
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1. In a packet telephony communication system, a method for handing audio data for output to a packet network, comprising the steps of:
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a. selecting an interim data rate for processing audio information, wherein said interim rate is substantially higher than an output data rate to reduce latency in a buffer;
b. converting audio information to audio data having a data rate equal to the selected interim data rate;
c. buffering in said buffer the converted audio data stream; and
d. downsampling the data rate of the buffered audio data stream to a rate equal to the output data rate, such flat performing said selecting, converting, buffering and downsampling steps results in reducing latency. - View Dependent Claims (2, 3, 4, 5, 6, 7)
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8. In a packet telephony communication system, a method for handling audio data input from a packet network, comprising the steps of:
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a. selecting an interim data rate for processing an audio data stream, wherein said interim rate is substantially higher than an input data rate to reduce latency in a buffer;
b. upsampling the data rate of the audio data stream to a rate equal to the selected interim data rate;
c. buffering in said buffer the upsampled audio data stream; and
d. processing the buffered audio data at the selected interim data rate, such that performing said selecting, upsampling, buffering and processing steps results in reducing latency. - View Dependent Claims (9, 10, 11, 12, 13, 14)
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15. In a packet telephony communication system, a method for handling audio data for output to or input from a packet network, comprising the steps of:
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a. processing audio information for output to the packet network, including the substeps of;
i. selecting a first interim data rate for processing the audio information for output to the packet network, wherein said first interim data rate is substantially higher than an output data rate to reduce latency in a first buffer;
ii. converting audio information to a first audio data stream having a data rate equal to the selected first interim data rate;
iii. buffering in said first buffer the first audio data stream; and
iv. downsampling the data rate of the buffered first audio data stream to a rate equal to the output data rate; and
b. processing audio information input from the packet network, including the substeps of;
i. selecting a second interim data rate for processing the audio information input from the packet network, wherein said second interim data rate is substantially higher than an input data rate;
ii. upsampling the data rate of a second audio data stream to a rate equal to the selected second interim data rate to reduce latency in a second buffer;
iii. buffering in said second buffer the upsampled second audio data stream; and
iv. processing the buffered second audio data stream at the selected second interim data rate such that said processing audion information for output and said processing audio information input reduces latency. - View Dependent Claims (16, 17, 18, 19, 20, 21, 22, 23, 24)
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25. An article of manufacture comprising a computer-readable medium having stored thereon instructions for use in conjunction with a computer-based packet telepony communication system, said instructions which, when executed, define a series of steps causing the packet telephony communication system to reduce latency in handling audio data for output to a packet network, said steps comprising:
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a. converting audio information to audio data having a selected interim data rate, wherein said selected interim rate is substantially higher than an output data rate to reduce latency in a buffer;
b. buffering in said buffer the converted audio data stream; and
c. downsampling the data rate of the buffered audio data stream to a rate equal to the output data rate, such that said converting, buffering and downsampling steps reduce latency. - View Dependent Claims (26, 27, 28, 29, 30, 31)
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32. An article of manufacture comprising a computer-readable medium having stored thereon instructions for use in conjunction with a computer-based packet telephony communication system, said instructions which, when executed, define a series of steps causing the packet telephony communication system to reduce latency in handling audio data input from a packet network, said steps comprising:
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a. upsampling the data rate of an audio data stream to a rate equal to a selected interim data rate, wherein said interim rate is substantially higher than an input data rate to reduce latency in a buffer;
b. buffering in said buffer the upsampled audio data stream; and
c. processing the buffered audio data at the selected interim data rate, such that said upsampling, buffering and processing steps reduce latency. - View Dependent Claims (33, 34, 35, 36, 37, 38)
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39. An article of manufacture comprising a computer-readable medium having stored thereon instructions for use in conjunction with a computer based packet telephony communication system, said instructions which, when excuted, define a series of steps causing the packet telephony communication system to reduce latency in handling audio data for output to or input from a packet network, said steps comprising:
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a. processing audio information for output to the packet network, including the substeps of;
i. selecting a first interim data rate for processing the audio information for output to the packet network wherein said first interim data rate is substantially higher thin an output data rate to reduce latency in a first buffer;
ii. converting audio information to a first audio data stream having a data rate equal to the selected first interim data rate;
iii. buffering in said first buffer the first audio data stream; and
iv. downsampling the data rate of the buffered first audio data stream to a rate equal to the output data rate; and
b. processing audio information input from the packet network, including the substeps of;
i. selecting a second interim data rate for processing the audio information input from the packet network, wherein said second interim data rate is substantially higher than an input data rate to reduce latency in a second buffer;
ii. upsampling the data rate of a second audio data stream to a rate equal to the selected second interim data rate;
iii. buffering in said second buffer the upsampled second audio data stream; and
iv. processing the buffered second audio data stream at the selected second interim data rate, such that said processing audio information for output and said processing audio information input reduces latency. - View Dependent Claims (40, 41, 42, 43, 44, 45, 46, 47, 48)
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Specification
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Current AssigneeAT&T Corporation (AT&T, Inc.)
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Original AssigneeAT&T Corporation (AT&T, Inc.)
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InventorsLyons, Robert Patrick, Robinson, Bethany Scott, Katseff, Howard Paul
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Primary Examiner(s)Kizou, Hassan
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Assistant Examiner(s)NGUYEN, HANH N
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Application NumberUS08/985,226Time in Patent Office1,972 DaysField of Search370/352, 370/412, 370/389, 370/428, 370/429, 370/356, 370/545, 370/353, 370/354, 370/355, 375/355, 375/372, 375/231, 364/400.01, 704/200, 704/201, 704/203, 704/219, 704/262US Class Current370/352CPC Class CodesH04L 25/05 Electric or magnetic storag...H04M 7/006 Networks other than PSTN/IS...H04M 7/0072 Speech codec negotiation H0...
