FBE200 audio issue and feature request


Currently I have a 48kHz input through HDMI, if I enable resample or volume adjustment the audio quality gets really bad.

Would it be possible to add an audio delay adjustment, both positive and negative to compensate for AV input synchronization issues. Could this adjustment be selected in both ms and frames (based on input frame rate)?


My friend:

Our engineers are also looking for the cause of this problem, but not clear of the cause.
Do you have any more in-depth research and discovery?

have a good day.


If you’re using ffmpeg or libavutil then it may be a problem when converting between planar and interleaved audio streams.


Maybe the task of resampling audio shouldn’t be done by the encoder at all. The user should simply ensure that proper audio format comes already prepared from the video source. Any kind of resampling will sooner or later lead do audio quality degradation, either done by this encoder or some other piece of equipment.

Sample rate and audio volume should only specified when using analog Line Input, when using HDMI for audio, re-sampling, voume setting etc. should be disabled imho. This also probably causes too much stress on the encoder’s CPU, which may lead to other problems later during operation.

Maybe a single option would be needed for HDMI audio: downmiix to stereo, for the cases when 5.1 or 7.1 audio comes with the HDMI signal, to downmix that to 2.0 audio - but that shouldn’t require resampling either.

If HDMI audio is used, the encoder should simply create the encoded live stream at the audio sample rate which is detected at HDMI input, that’s all.


I disagree, the encoder should have at least basic audio resampling and volume level adjustment. PCM audio downmix and dolby digital input (DTS, etc.) would be a nice to have, maybe even an option of passing the audio input bitstream straight out without re-encoding. Sometimes you’re stuck using a device that you can not change the output parameters to match what you’d like to stream (which is what is my problem is). Internet streams usually use 44.1 but most OTA boxes output 48 as that’s a normal broadcast rate (at least here in ATSC land).


I look this problem vith audio IN. Sound stream push ahead video.
Need latency ajustement between Audio and video streams + and -