Stream adaptation for latency
A system and method for adapting an audios stream for reducing latency. The method may include the steps of, and the system may function to, receive an audio stream having a packet buffer and an audio buffer, measure the audio buffer depth of the audio buffer, measure the presentation time margin of at the input to the packet buffer, and determine an adaptation level for latency based on the measured values.
1. A method for adapting an audio stream to reduce latency, said method comprising the steps of:
(a) receiving an audio stream at an audio output, said audio output having a packet buffer and an audio buffer;
(b) measuring an audio buffer depth of said audio buffer;
(c) measuring a presentation time margin at an input to the packet buffer;
(d) comparing said audio buffer depth to a depth limit, and said presentation time margin to a time limit; and
(e) determining an adaptation level based on said comparing.
2. The method of claim 1 , further comprising the step of determining at least one adaptation level selected from the group comprising Level 0, Level 1, Level 2, Level 3, and Level 4.
3. The method of claim 2 wherein if the measured audio buffer depth is greater than said depth limit and said measured presentation time limit is greater than said time limit, then adaptation Level 0 is undertaken.
4. The method of claim 2 wherein if the measured audio buffer depth is greater than said depth limit and said measured presentation time limit is less than said time limit, then adaptation Level 1 is undertaken.
5. The method of claim 2 wherein if the measured audio buffer depth is less than said depth limit and said measured presentation time limit is greater than said time limit, then adaptation Level 2 is undertaken.
6. The method of claim 2 wherein if the measured audio buffer depth is less than said depth limit and said measured presentation time limit is less than said time limit, then adaptation Level 3 is undertaken.
7. The method of claim 6 wherein adaptation Level 4 is undertaken once adaptation Level 3 is undertaken for more than one second.
8. The method of claim 1 wherein said determining of said adaptation level includes the step of dividing said audio stream into at least a low stream and a high stream, wherein said low stream is transmitted at a lowest data rate and said high stream is transmitted at a data rate higher than said lowest data rate.
9. The method of claim 8 wherein said determining an adaptation is based on determining the following adaptation levels:
Level 0: Normal, no adaptation required;
Level 1: Lower a bit depth of said low stream audio to 16 bits; and
Level 2: Adaptation of Level 1, plus increasing a PHY rate of said low stream and said high stream.
10. The method of claim 8 wherein said determining an adaptation is based on determining the following adaptation levels:
Level 0: Normal, no adaptation required; and
Level 1: Using an audio compression algorithm, reduce a number of bits of said high stream.
11. The method of claim 9 , wherein said diving of said audio streams include four levels of audio stream, Tranche 0, Tranche 1, Tranche 2, and Tranche 3, such that Tranche 0 is the lowest stream and Tranche 3 is the highest stream, and said determining of said adaptation is based on the following adaptation levels:
Level 0: Normal
Level 1: Lowering said bit depth of Tranche 1 audio to 16 bits and discard Tranche 2 and Tranche 3;
Level 2: Level 1 plus increasing the PHY Rate of Tranche 0 and Tranche 1;
Level 3: Level 2 plus decimating Tranche 1; and
Level 4: Level 3 plus mixing down; 7.1 to 5.1 to 3.1 to 2.0.
12. The method of claim 9 wherein said time limit and said depth limit are calculated using the following relationships:
time limit=200%*Interleaving Block Size
depth limit=audio buffer size−(200%*Interleaving Block Size).
13. The method of claim 1 wherein the said step of measuring said presentation time margin is measured using the following equation:
PTM
(
k
)
=
1
N
∑
n
=
0
N
-
1
[
PTime
(
k
-
n
)
-
TSF
(
k
-
n
)
]
where k and n are indexes at interleaving block times.
14. The method of claim 1 wherein the audio buffer is an ALSA buffer.
15. The method of claim 1 wherein the packet buffer is a UDP buffer.
16. A system for adapting an audio stream to reduce latency, said system comprising:
(a) an audio input component capable of receiving an audio stream from a source and transmitting the audio stream, said audio stream having a bit rate;
(b) an audio output component capable of receiving the audio stream transmitted by the audio input component, and including a packet buffer and an audio buffer;
(c) a computing component using a network capacity metric based on the statistics of the difference between the time the audio stream is to be played and when the audio data stream is received at the audio output component, and
(d) if a reduced network capacity exists, employing an adaptation component to alter the bit rate of the audio stream to a constant latency by dividing the audio stream into at least a low stream and a high stream, wherein said low stream is transmitted at a lowest data rate and said high stream is transmitted at a data rate higher than said lowest data rate, and wherein an adaptation employed by the adaptation component is determined by comparing an audio buffer depth to a depth limit and a presentation time margin to a time limit.
17. The system of claim 16 , wherein said packet buffer comprises a UDP buffer.
18. The system of claim 16 , wherein said audio buffer comprises an ALSA buffer.
19. The system of claim 16 wherein said audio input component includes a WiFi Buffer component.
20. The system of claim 16 wherein said audio input component includes an interleaving and packetizing element.
21. The system of claim 16 , wherein said audio output component includes a Wifi Buffer component.
22. The system of claim 16 wherein said audio output component includes a de-interleaving, concealing, and SRC element.