AAC MPEG-4

Torben is right. You should use he for 64 kbps.

This is what we use

(directory where enc_aacPlus.ex is located) …\enc_aacPlus.exe - - --br 64000 --he --silent --rawpcm 44100 2 16
(directory where enc_aacPlus.ex is located) …\enc_aacPlus.exe - - --br 32000 --he --silent --rawpcm 44100 2 16

I found this somewhere on the internet:

--lc             : Encode as LC-AAC (default)
                      (mono:8-16, stereo: 16-320, 6ch: 160-320 kbps)
--he           : Encode as HE-AAC
                       (mono:8-64, stereo: 16-128, 6ch: 96-213 kbps)
--high        : Encode as HE-AAC+ with hight bitrates
                       (mono:8-160, stereo: 96-256, 6ch: 8-256 kbps)

So, I suppose you can use ‘high’ when your samplerate is higher than 96 kbps.

I hope you can get it to work.

I put it like this:

In encoding option: C:\Program Files (x86)\mAirList 5.3\enc_aacPlus.exe - - --br 64000 --he --silent --rawpcm 44100 2 16

And I downloaded Enc_aacPlus.exe here : https://aacpluscli.codeplex.com/releases/view/37963 which I put at the root of mairlist.

It does not work :confused:


aac.png

ok thanks i have change to enc_aacPlus.exe - - --br 64000 --he --silent --rawpcm 44100 2 16

But look, i get error 471 on the Shoutcast server and there for not listed on Shoutcast.com


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2.PNG

Don’t forget to enter the MIME type in the field that appeared next to the Format field: audio/aacp

Torben Super thanks

Now it all works and is listed at Shoutcast


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Disclaimer: Use of AAC encoder might be protected by software patents in some countries; that’s why mAirList does not support it officially. Use at your own discretion.

Is it possible to get this to work with the integrated streaming encoder, please? What would you need to change to be able to listen to the stream?

Have you tried it? Actually the settings should be the same. Never tested it though.

Yes and no is the true answer. I set up the encoder and it seems to be streaming (i.e. no errors), but I don’t seem to be able to listen to it, however. With an mp3 stream you just enter the local ip address into VLC or winamp and away it goes, but not with this.

the encoder is bad try an other one.

Which one would you prefer?

I tried it and it works without any problem.

Sorry, I would have to disagree. It works well with shoutcast, icecast etc, and the integrated encoder is useful to stream from a remote spot to the studio. My question was designed to see if you could cut down on the data used i.e. comparing a 128kb mp3 file to say a 48kb aac+ one.

I just noted a small thing here

See my image it say 22.05 kHz but in encoder there is 44100 kHz, anyone know way?

C:\Program Files (x86)\mAirList 5.3\enc_aacPlus.exe - - --br 64000 --he --silent --rawpcm 44100 2 16 Is the current using setting


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enc.PNG

You did not set the MIME type (in the blank field at the right of ‘Audio Format’) to “audio/aacp”. Maybe that’s the problem?

44100 is the input sample rate. The parameter is to tell the command line encoder the format and bitrate of the raw PCM samples coming in via stdin.

I presume the audio is always resampled to 22050 as the output frequency when using low bitrates like 64kbps.

By the ways, has anyone tried the FDK encoder? It supports stdin and stdout, so it should be compatible. Just can’t find any Windows binary for it right now.

https://github.com/nu774/fdkaac

[quote=“Torben, post:23, topic:11042”]By the ways, has anyone tried the FDK encoder? It supports stdin and stdout, so it should be compatible. Just can’t find any Windows binary for it right now.

https://github.com/nu774/fdkaac[/quote]
I have used it on Linux with Liquidsoap, but I assume this is not helpful due to it’s native implementation to Liquidsoap. This is actuall driving our mobile streams.

[quote=“Torben, post:22, topic:11042”]44100 is the input sample rate. The parameter is to tell the command line encoder the format and bitrate of the raw PCM samples coming in via stdin.

I presume the audio is always resampled to 22050 as the output frequency when using low bitrates like 64kbps.[/quote]

No, it is not resampled to 22050. I have 64kbps streams with the same settings : … (location of enc_aacPlus.exe)\enc_aacPlus.exe - - --br 64000 --he --silent --rawpcm 44100 2 16
and the result is a stream at 44.1 KHz and 64 kbps. Maybe there is something wrong with richjoa’s Shoutcast server settings.

I did a fresh install on on Linux hard to see what should be wrong there.

[quote=“CROOZEmaster, post:25, topic:11042”][quote=“Torben, post:22, topic:11042”]44100 is the input sample rate. The parameter is to tell the command line encoder the format and bitrate of the raw PCM samples coming in via stdin.

I presume the audio is always resampled to 22050 as the output frequency when using low bitrates like 64kbps.[/quote]

No, it is not resampled to 22050. I have 64kbps streams with the same settings : … (location of enc_aacPlus.exe)\enc_aacPlus.exe - - --br 64000 --he --silent --rawpcm 44100 2 16
and the result is a stream at 44.1 KHz and 64 kbps. Maybe there is something wrong with richjoa’s Shoutcast server settings.[/quote]