Wegener Communications 6420 User Manual
iPump 6420 User’s Manual
www.wegener.com
800070-01 Rev B
Chapter 3, Page 71
2. Audio Language descriptor (select Language Descriptor from list as found in the
PMT, or use the first available in the PMT as indicated by the ‘
*
’ wildcard)
Codec, resamplers, timing adjustments (buffer-locked loop)
The live audio PES stream is decomposed into separate audio ES frames, and the
The live audio PES stream is decomposed into separate audio ES frames, and the
compressed data is passed to a software codec. In the i6420, an award-winning, industry-
standard audio codec is used. The resulting linear data is passed to an output buffer to prepare
for further processing.
standard audio codec is used. The resulting linear data is passed to an output buffer to prepare
for further processing.
At this point, the linear audio samples are output by the codec at the sample rate used within
the uplink compression system. The i6420, however, does not use the PCR or PTS timing
signals normally conveyed in DVB/MPEG Transport streams. Instead, the output timing is set
by a fixed oscillator in the i6420. Now, this would normally cause the buffer capturing the data
from the codec to eventually over or under-flow, since the uplink is creating and the i6420 audio
output consuming the audio samples at different rates. So the i6420 maintains a (fairly) constant
buffer by dropping or repeating samples. This is done with an innovative algorithm which seeks
out periods of low-complexity audio, either quiet moments or simple tones. Then samples are
dropped and added in groups which neatly match the cycle period. Thus, samples are dropped or
repeated less often, and when it is done, it is hidden in such a way as to render it inaudible to
even professional listeners.
signals normally conveyed in DVB/MPEG Transport streams. Instead, the output timing is set
by a fixed oscillator in the i6420. Now, this would normally cause the buffer capturing the data
from the codec to eventually over or under-flow, since the uplink is creating and the i6420 audio
output consuming the audio samples at different rates. So the i6420 maintains a (fairly) constant
buffer by dropping or repeating samples. This is done with an innovative algorithm which seeks
out periods of low-complexity audio, either quiet moments or simple tones. Then samples are
dropped and added in groups which neatly match the cycle period. Thus, samples are dropped or
repeated less often, and when it is done, it is hidden in such a way as to render it inaudible to
even professional listeners.
After this step, the audio data stream must be passed to an audio mixer where it may be
summed with the outputs of codec stages which have processed audio files. The mixer must
output the audio samples at a user-set sample-rate, so it requires all its inputs to be the same rate.
The software supplies re-sampling to all mixer inputs as needed. Again, this is performed by an
industry-standard 3
output the audio samples at a user-set sample-rate, so it requires all its inputs to be the same rate.
The software supplies re-sampling to all mixer inputs as needed. Again, this is performed by an
industry-standard 3
rd
party software module.
The user controls, for each Audio Decoder (Port) are:
1. Output audio sample rate
Audio Buffer delay
User should be aware that the decoded live audio is delayed in an audio buffer with a
User should be aware that the decoded live audio is delayed in an audio buffer with a
nominal factory-set depth of 500 mS. As file audio is pulled and decoded, it too is passed to the
same buffer and encounters the same delay. This must be taken into account when constructing
an overall system timing model (see Section 3.4.3).
same buffer and encounters the same delay. This must be taken into account when constructing
an overall system timing model (see Section 3.4.3).
The user controls are:
1. Audio buffer delay (factory set, but may be adjusted with debug access).