Controlling A Jitter Buffer
US-2015350099-A1 · Dec 3, 2015 · US
US9626985B2 · US · B2
| Field | Value |
|---|---|
| Publication number | US-9626985-B2 |
| Application number | US-201415036926-A |
| Country | US |
| Kind code | B2 |
| Filing date | Oct 21, 2014 |
| Priority date | Nov 15, 2013 |
| Publication date | Apr 18, 2017 |
| Grant date | Apr 18, 2017 |
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Audio processing methods and apparatus are provided. An audio processing method may include: receiving audio data packets; buffering the audio data packets to a buffer; reading the audio data packets from the buffer and playing the audio data packets; accumulating an actual total playing time length and a total sampling time length of the audio data packets that currently have been read from the buffer and have been played; and suspending reading and playing, when a sum of sampling time lengths of audio data packets that are buffered and unread in the buffer is less than or equal to a first threshold, until a sum of sampling time lengths of the audio data packets in the buffer that are unread is greater than or equal to a current network jitter estimated value.
Opening claim text (preview).
What is claimed is: 1. An audio processing method, comprising: receiving audio data packets; buffering the audio data packets to a buffer; reading the audio data packets from the buffer and playing the audio data packets; accumulating an actual total playing time length and a total sampling time length of the audio data packets that currently have been read from the buffer and have been played; and suspending reading and playing, when a current sum of sampling time lengths of the audio data packets that are buffered and unread in the buffer is less than or equal to a first threshold, until a sum of sampling time lengths of the audio data packets in the buffer that are unread is greater than or equal to a current network jitter estimated value, the current network jitter estimated value being obtained based on the accumulated actual total playing time length and the accumulated total sampling time length. 2. The audio processing method according to claim 1 , wherein the step of reading includes: reading the audio data packets from the buffer according to time sequence numbers. 3. The audio processing method according to claim 1 , wherein the step of accumulating includes: accumulating the actual total playing time length within a pre-determined time period from a preset moment to a current moment, and accumulating the total sampling time length of the audio data packets that have been read from the buffer and have been played within the pre-determined time period. 4. The method according to claim 1 , wherein: the step of buffering includes: buffering the audio data packets of a first audio file to the buffer; and the step of accumulating includes: accumulating an actual total playing time length of the first audio file and a total sampling time length of the audio data packets of the first audio file that currently have been read from the buffer and played. 5. The method according to claim 1 , wherein: the current network jitter estimated value equals to a difference between the accumulated actual total playing time length and the accumulated total sampling time length. 6. The method according to claim 1 , wherein: the current network jitter estimated value equals to a multiplication of a jitter risk coefficient β and a difference between the accumulated actual total playing time length and the accumulated total sampling time length. 7. The method according to claim 6 , wherein: the jitter risk coefficient β ranges from 0.8 to 1.5. 8. The audio processing method according to claim 6 , wherein the first threshold equals to 0 second. 9. A terminal device, comprising: a receiving and buffering unit, configured to receive audio data packets and buffer the audio data packets to a buffer; a reading and playing unit, configured to read the audio data packets from the buffer and play the audio data packets; an accumulating unit, configured to accumulate an actual total playing time length and a total sampling time length of the audio data packets that currently have been read from the buffer and have been played; and a buffering and playing unit, configured to suspend reading and playing when a current sum of sampling time lengths of the audio data packets that are buffered and unread in the buffer is less than or equal to a first threshold, until a sum of sampling time lengths of the audio data packets in the buffer that are unread is greater than or equal to a current network jitter estimated value, the current network jitter estimated value being obtained based on the accumulated actual total playing time length and the accumulated total sampling time length. 10. The terminal device according to claim 9 , wherein the reading and playing unit is further configured to read the audio data packets from the buffer according to time sequence numbers. 11. The terminal device according to claim 9 , wherein the accumulating unit is specifically configured to accumulate the actual total playing time length within a pre-determined time period from a preset moment to a current moment and to accumulate the total sampling time length of the audio data packets that have been read from the buffer and have been played within the pre-determined time period. 12. The terminal device according to claim 9 , wherein the receiving and buffering unit is specifically configured to buffer the audio data packets of a first audio file to the buffer; and the accumulating unit is specifically configured to accumulate an actual total playing time length of the first audio file and a total sampling time length of the audio data packets of the first audio file that currently have been read from the buffer and played. 13. The terminal device according to claim 9 , wherein the current network jitter estimated value equals to a difference between the accumulated actual total playing time length and the accumulated total sampling time length; or the current network jitter estimated value equals to a multiplication of a jitter risk coefficient β and a difference between the accumulated actual total playing time length and the accumulated total sampling time length. 14. The terminal device according to claim 13 , wherein: the jitter risk coefficient β ranges from 0.8 to 1.5.
Flow control; Congestion control · CPC title
involving operations for analysing the audio stream, e.g. detecting features or characteristics in audio streams (arrangements characterised by components specially adapted for monitoring, identification or recognition of audio in broadcast systems H04H60/58) · CPC title
Electricity · mapped topic
Processing of audio elementary streams · CPC title
Jitter · CPC title
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