RTP Buffering - Frame Based Buffering

Introduction

In Song module version 8 a new RTP buffering method called frame based buffering was introduced. The algorithm calculates the audio buffer level in milliseconds rather than in bytes.

Features

Frame based buffering allows:

• configurable decoding delay with one frame accuracy
• synchronisation of several decoders to the same stream (just by configuring them to the same initial delay)
• stable delay over long period of time
• automatic correction of clock difference between encoder and decoder

Applications

The following applications use frame based buffering:

Application Name	Version
Streaming Client	2.17
Annuncicom Full Duplex	0.21
RTP STL	2.01

Configuration

The only configuration parameter for the RTP decoder is the delay in milliseconds.

The delay parameter is the desired processing delay of the decoder (between the network input and the audio output). Please note that the end-to-end delay between the encoder and the decoder might be (significantly) different to the value configured.

In an ideal case the delay parameter would be 0 ms, however due to device's internal buffers a small delay (depending on the hardware) is inevitable. The delay value should also cover possible temporary network hick-ups (jitter). E.g. if the network might sometimes delay the packet delivery by 20ms due to a temporary load, the configured parameter should not be less than 20ms.

The maximum configurable delay is limited by the device's internal buffer (64, 32 or 16kB).

The following sections show recommended delay values for various audio formats.

Recommended Settings

MP3

The following table shows the minimum and the maximum possible delay with MP3 constant bitrate. The maximum delay differs between the Streaming Client, which has 64kB audio buffer available, and ABCL (Annuncicom FDX, STL), which features only 32kB buffer. The minimum delay includes 100ms network jitter.

MP3 Format	Min delay	Max delay (SC)	Max delay (ABCL)
CBR 320kbps	150ms	1,588ms	769ms
CBR 256kbps	163ms	2,011ms	987ms
CBR 192kbps	183ms	2,741ms	1,349ms
CBR 160kbps	200ms	3,277ms	1,638ms
CBR 128kbps	225ms	4,121ms	2,073ms
CBR 64kbps	350ms	8,342ms	4,246ms
CBR 32kbps	600ms	16,784ms	8,592ms

Multiple Device Synchronisation

Multiple devices receiving the same RTP stream can be configured to play in sync by entering the same delay parameter.

Barix recommends to use broadcast or multicast together with synchronisation, otherwise a small inaccuracy (few milliseconds) might be caused by the network delivery to different locations.

Deliberate Delays

In some applications it is desired to artificially delay the audio. E.g. in a tunnel to eliminate the delay caused by the distance between the devices.

An artificial delay can be introduced by configuring the devices to different delay values. E.g. 100ms, 120ms, 140ms, 160ms, etc.