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Evaluating Mezzmo, couple of questions
Hi,
i'm currently trying out Mezzmo as Media server. So far it works quite well, but I do have a few issues:
- when files get transcoded to H264 format, it seems as if streaming doesnt start until transcoding is complete. Once the file is 100% transcoded I can view it ok...
- Transcoding HD ts files causes audio / video to go out of sync. Or at leat so it seems on the TV. These are .ts files I've taken from my Humax decoder.
- Itunes music files (m4a) plays 'too slow' (like playing a 45rpm record at 33rpm)... Other music (mp3) plays fine.
I'm currently streaming this to my tv (Philips PFL7605) using the Philips 8000 series profile.
TV and MediaServer are connected using a 200Mbs HomePlug.
Are there any settings I need to changes to get these things working correctly?
Tnx in advance...
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I found out what the problem was with the m4a files. They were being transcoded to something the TV didnt quite understand (or implemented correctly...). I change the profile so that m4a now get transcoded to mp3 files. New problem with this is that the bitrate is too low. I got it to transcode to a fix bitrate of 192K by using:
<ffmpegadditional>-ab 192k</ffmpegadditional>
Is there a way to use to original bitrate of the m4a file?
Alsop, the audio sync issue seems to be cased by the TV and/or the network. When opening the Transcoded file in VLC-player, audio is fine...
Any clues as to what is going on?
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Also, when I fast forward in a movie being streamed, the 'steps' are rather small. Is there a way to make FastForward 'faster'? Meaning, stepping ahead 1 minute instead of a sec? (or something like that?).
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You have to hold key down
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Can you please send a message to support (at) conceiva (dot) com and we'll send you an updated device profile and also an updated version of FFmpeg that may fix some of the transcoding issues.
The original rate of the m4a should be used if you didn't specify any <bitrate> in the MP3 container. What does your current MP3 transcoding container look like?
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my current mp3 container looks like:
<avcontainer id="mp3" name="mp3" encode="1" mimetype="audio/mp3" baseline="1" dlnaprofile="MP3">
<audio>
<format id="mp3" encode="1" codec="libmp3lame">
</format>
</audio>
<ffmpegadditional>-ab 192k</ffmpegadditional>
</avcontainer>
So basically i just enabled the 'encode' flags and added the ffmepadditional -ab setting. Otherwise FFmpeg uses the default -ab which is 64k...
I also disabled the transcoding to LPCM as that seemed to be causing the problem in the first place, while original mp3 files played ok.
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