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Just tested it with Foobar and here it works. I have used foobar 1.3.9 though (latest version). The other components have the same version numbers as yours.
Conversion is however limited to the 48kHz core stream.

Should it not work, feel free to PM me a link to the file and I'll have a closer look.

Hi all!! Could be possible to get sacd support on audio muxer? i need to decode by foobar and send huge wav's to audiomuxer...

native decoding expected?

Thanks in advance!!

Hi! I'm not sure I get it right that Pl4yitLOUD is the author of AudioMuxer. First of all thanks for your great program. It's been a great time saver in many occasions.

I use it to, among other things, encapsulate in FLAC files DTS and AC3 streams, for having multichannel playback with my Squeezeboxes (that have bitperfect output through SPDIF). Now I find myself in a strange situation, because Squeezeboxes can be synched throughout your hose (I have several in different rooms) but I have only one connected to an AV receiver. If any of them is synched to the AV Receiver connected one... it's white noise madness.

The Squeezebox Server allows for customized transcoding and, basically, I'd love to set up things so that if playback of one of this FLAC/DTS-AC3 files is initiated on a "non AV Receiver connected" unit, the file is transcoded to a stereo version, downmixed.

Now I've isolated the procedure that allows me to get an mp3 or flac two channels downmixed file but it involves three steps. First I need to decode the flac to wav, then I have to feed the resulting wav to http://www.ac3filter.net/wiki/AC3Filter_tools#bsconvert BSconvert] and finally, through FFMPEG I can convert the resulting .dts/.ac3 file to a downmixed stereo audio file (either flac or mp3 or whatever basically). The problem is that, as far as I know, the Squeezebox server allows only for two pass transcoding and it needs to happen through stdin and stdout.

BSconvert comes from the same suit of AC3 Filter Tools you uses spdifer from. Hence my question: are you aware of any way I could transform the above process in a two step one, using stdin and stdout?

I provide you with an example of how, for instance, Squeezebox Server handles the conversion of mp4 (aac) to mp3 to give you an idea, it's a line in a .conf file editable by the end user.

[faad] -q -w -f 1 $START$ $END$ $FILE$ | [lame] --silent -q $QUALITY$ $BITRATE$ - -

as you can see faad is used with the -w option to write to stdout and lame gets that output and does its job.

This isn't strictly audiomuxer related (unless I'm wrong and you provide a commandline interface) but if there is anybody in the world that could have an idea about this, I think it's you. As, basically, BSconvert is capable of reversing what spdifer does.

Thanks for reading. It's all for surround goodness.

Damn... I hope you didn't lose too much time. I came to the same conclusions after checking this morning here and deciding to keep on banging my head against the wall...

http://forums.slimdevices.com/showthread.php?19260-Playing-AC3-DTS-tracks-sourced-from-DVD&p=848101&viewfull=1#post848101

I remembered about my post here just now and wanted to say not to lose time on it...

The trick, as you found out, was to have FLAC decoding the file to stdout... who knows what FFMPEG was doing internally when trying to do it in one step. I thought I needed BSconvert when I tried doing it in one step in FFMPEG and it wasn't working. If it's not decoded beforehand it treats the flac file as a normal wave and re-encodes just noise.

Again, thank you for your awesome program, thank you for taking the time to help me. And sorry for having wasted some of your time.

No problem. Glad that it is all sorted out now.

What do you mean by including "he-aac" options? Do you refer to the possibility to also encode to AAC+ for everything up to 48kHz, since for now AAC is used as the default because all various channel assignments are available in that format. AAC+ is more restricted to the more common ones (mono, stereo, 4.0, 5.1 & 7.1).