Jump to: navigation, search

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 sometimes delays 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).

Recommended Settings

The following table lists recommended delay values for various audio formats. The value includes 2-frame jitter and is independent on hardware/software.

Audio format Delay
MP3 600ms
uLaw/ALaw 8kHz mono 444ms
PCM 8kHz mono 444ms
uLaw/ALaw 12kHz mono 316ms
PCM 12kHz mono 316ms
uLaw/ALaw 24kHz mono 188ms
PCM 24kHz mono 188ms
uLaw/ALaw 32kHz mono 156ms
PCM 32kHz mono 152ms
PCM 44.1kHz mono 110ms
PCM 44.1kHz stereo 79ms
PCM 48kHz stereo 72ms

Maximum Settings

This section explains the minimum and the maximum delay values for different audio formats and platforms.

The hardware is divided into two groups:

  • Micronas (MAS) based devices: Annuncicom 100/155/200/1000, Exstreamer 1000
  • VLSI based devices: Exstreamer 100/110/200

MP3 CBR

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

MP3 VBR and ABR

Variable or average bitrate the minimum and delay depends on the bitrate variation interval. The minimum delay is taken from the CBR table for the low end of the interval, whereas the maximum delay is the CBR value for the high end of the interval.

Please note that most MP3 encoders use the whole bitrate range starting from the lowest bitrate 32kbps. E.g. VBR 128kbps varies from 32 to 128kbps

MP3 Format Min delay Max delay (SC) Max delay (ABCL)
32-320kbps 600ms 1,588ms 769ms
32-256kbps 600ms 2,011ms 987ms
32-192kbps 600ms 2,741ms 1,349ms
32-160kbps 600ms 3,277ms 1,638ms
32-128kbps 600ms 4,121ms 2,073ms
32-64kbps 600ms 8,342ms 4,246ms

PCM

In uncompressed audio (PCM, uLaw or ALaw) the minimum and maximum delay depend on the bit rate and on the hardware.

The following table lists minimum and maximum settings for all standard RTP audio formats:

Format Min delay MAS Min delay VLSI Max delay (SC) Max delay (ABCL) Max delay (ABCL full duplex)
uLaw 8kHz mono

ALaw 8kHz mono

80ms 424ms 8171ms 4075ms 2027ms
PCM 8kHz mono 60ms 424ms 4075ms 2027ms 1003ms
uLaw 12kHz mono

ALaw 12kHz mono

67ms 296ms 5441ms 2710ms 1345ms
PCM 12kHz mono 54ms 296ms 2710ms 1345ms 662ms
uLaw 24kHz mono

ALaw 24kHz mono

54ms 168ms 2710ms 1345ms 662ms
PCM 24kHz mono 47ms 168ms 1345ms 662ms 321ms
uLaw 32kHz mono

ALaw 32kHz mono

50ms 136ms 2027ms 1003ms 491ms
PCM 32kHz mono 43ms 134ms 1005ms 493ms 237ms
PCM 44.1kHz mono 31ms 97ms 729ms 358ms 172ms
PCM 44.1kHz stereo 16ms 72ms 364ms 179ms 86ms
PCM 48kHz stereo 15ms 66ms 335ms 164ms 79ms

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.

Back to Main_Page#Software_Topics