Modifying the latency and jitter of audio/video streams

Important:If a player is using firmware version 6.2, the method described below requires firmware version 6.2.147 or later.

You can modify the amount of time it takes for audio/video streams to begin playback by formatting the stream URL as follows: <streaming_url>?latency=<value_in_ms>&jitter=<value_in_ms>

Examplehttp://www.example.com/stream1?latency=-450&jitter=50

The default latency value is 0 milliseconds and the default jitter value is 300 milliseconds, which allows the audio- and video-stream buffers to fill up before playback begins; this makes playback less likely to stutter when the stream has a high-bitrate or the network is slow. Using smaller latency and jitter values will not change the buffer size. Instead, it will give the buffers less time to fill up before playback begins. We recommend thoroughly testing custom latency values in a target network environment before deployment.

The "latency" value is measured as a deviation from the default latency (in milliseconds). For example, a value of -250 reduces the latency by a quarter of a second, while a value of 0 specifies the default latency. Tests have shown that usable latency values extend down to approximately -450ms (though this value may differ depending on the network environment)--while jitter values can be reduced to approximately 30ms. Reducing the latency/jitter too much will result in obvious playback stutter.

You can also increase the latency value beyond 0. This will increase the size of the streaming buffer and delay playback. Increasing latency to 10 seconds or more will likely require use of the vcdbsize and acdbsize parameters to increase the memory allocated to the buffer.

Have more questions? Submit a request

0 Comments

Please sign in to leave a comment.
Can't find what looking for? Try to
Powered by Zendesk