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[radio42]

Hi,

a new verison of the enc_aacPlus command-line encoder is available!

You all know, that "BASSenc" can be used with any external command-line tool to encode sample data, if that tool supports STDIN to receive sample data and STDOUT, to receive back the encoded data.
So far there was no command-line tool available, which supported AACplus V2 encoding like this.
Now there is ;-)

Here is the download link: http://www.un4seen.com/filez/4/enc_aacPlus.zip

The trick is, that this tool uses the AACplus v2 encoder provided with Winamp 5.33 or above!
So if you install Winamp, you get this legally for free.

enc_aacPlus.exe was originally provided by Dmirty Alexandrov.
I just made a few adjustments (see below)...

NOTE!
"enc_aacplus.dll" and "nscrt.dll" must be copied into the directory of enc_aacPlus.exe!
Get these two dll's from your Winamp 5.33 or above plugins directory!

/*
* BIG FAT WARNING:
* this code based on "The Winamp Transcoder" plugin (http://www.srcf.ucam.org/~wdhf2/transcoder/)
*
* Copyright (c) 2005 Dmirty Alexandrov aka dimzon aka dimzon541 (dimzon541@gmail.com)
* Copyright (c) 2004 Will Fisher (will.fisher@gmail.com)
*
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
*    notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
*    notice, this list of conditions and the following disclaimer in the
*    documentation and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
*    derived from this software without specific prior permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
* NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
*
*
* This version is based on the aacPlusCLI project as published here: http://aacpluscli.codeplex.com/
* AACPlus v2 Encoder (using Winamp 5.33 or above enc_aacplus.dll and nscrt.dll)
* Coding Technologies encoder 8.0.3
* Build (9349): Mar 18 2008 at 09:00 AM
*
* The following enhancements have been made to allow this encoder to be used with BASS (www.un4seen.com):
* 1) "--silent" option added
* 2) "-" for STDOUT added
* 3) corrected wave chunk reading
*
*/

/*
Usage:
        enc_aacPlus.exe <wav_file> <bitstream_file> [options]

Input/Output:
        <wav_file>       : WAV input file (use - to read from STDIN)
                           (must be PCM 8 or 16 bit, 32000, 44100 or 48000 Hz)
        <bitstream_file> : Bitstream output file (use - to write to STDOUT)
                           (mp4box not possible, if STDOUT is selected)

Options:
        --br <bitrate>   : Set bitrate (CBR) in bps. Default is 128000
        --ps             : Enable Parametric Stereo (bitrates up to 56000, only with HE-AAC)
        --is             : Prefer Independed Stereo (disables Joint Stereo M/S coding)
        --mono           : Encode as Mono
        --mpeg2aac       : Force MPEG2 AAC (instead of let the encoder choose)
        --mpeg4aac       : Force MPEG4 AAC (instead of let the encoder choose)
        --lc             : Encode LC-AAC (bitrate up to 320000)
        --he             : Encode HE-AAC (bitrate up to 128000)
        --high           : Encode as HE-AAC+ with hight bitrates (bitrate up to 256000, multichannel not supported)
        --mp4box         : Wrap result into MPEG4 container (MP4BOX.EXE must be in the same directory)
        --mp4mux         : Output MPEG4 ISO compilant file .mp4 instead of raw .aac (MP4MUX.EXE must be in the same directory)
        --silent         : Set silent mode, no progress and other output
        --rawpcm <r c b> : Signal RAW PCM input intead of WAV
                 <r>     : Samplerate in Hz (32000, 44100 or 48000)
                 <c>     : Channels count (1 or 2 or above)
                 <b>     : Bit's per sample (8 or 16, little-endian)

Example:
enc_aacPlus.exe input.wav out.aac --br 56000
enc_aacPlus.exe input.wav out.m4a --br 32000 --he --mp4box
enc_aacPlus.exe - - --br 128000 --high --silent --rawpcm 44100 2 16
*/

[Sebastian_Mares]

I have no idea if the Nero encoder supports StdOut, but it certainly supports StdIn which should be enough if one only needs to encode data.

[radio42]

Yepp, that's true.
The nero Encoder tool only supports STDIN, which is sufficient for creating AACplus files.
But it does not support STDOUT, which is needed, if you want to get the encoded data back - e.g. if you want to stream the data to a SHOUTcast server ;-)

[Intimidated]

I would just like to say THANK YOU.

i've just spent hours trying to find an encoder with this functionality, and I should've looked on the BASS forums in the first place.

Thanks to BASS, I have now created a stream transcoder which has the capability of reading MP3, AAC/HE-AAC, Ogg Vorbis,  WMA and Ogg Flac, and can transcode into MP3 OGG and AAC Plus.

I cant thank you guys enough.

[radio42]

You're welcome

[KtoY]

Hi radio42, i want to know, how use enc_aacplus.dll without enc_aacplus.exe?

[radio42]

Since it is a regular Winamp plugin it must use the Winamp PlugIN API as documented by Winamp...

So each encoder implements an interface class called "AudioCoder".
You just need to query for the "CreateAudio3" process address in the dll, create an "AudioCoder" instance of that one and then call the "Encode" method for each chunk of data you are processing...

All the rest is (more or less) documented on the Winamp pages ;-)

[KtoY]

Thanks

[Mauricio Castro]

I am using as a basic source of example to mp3 and ogg. Let me a hint to make use of encryption to MP4 (aacp). Does anyone know what I say a command line?

///VEJA >> SEE

  // setup encoder command-line (raw PCM data to avoid length limit)
  if chk_Mp3.Checked then
  begin
    Content := BASS_ENCODE_TYPE_MP3;
    Comline := PChar(Format('lame -r -x -s 44100 -b %d -', [bitrate]));
  end
  else
  begin
 Content := BASS_ENCODE_TYPE_OGG;
     Comline := PChar(Format('oggenc -r -R 44100 -b %d -m %d -', [bitrate, 16]));
      // for better Quality on ogg just use -q (-2.0 - 10.0}

  end;

[Bert]

Dear Bernd,
i'm sorry, but what is new now?
I evaluate this for a long time now, my problem is that the encoder produces always a loud haerable click at the start of the recording.
Is this better now?

Best, Bert.

[radio42]

Hi Bert,

I might not fully understand your question?
a) "What is new now"?
-> To what are you referring - v2.4? enc_aacPlus.exe is there for quite some time and hasn't changed since then! It was just constantly adapted to always work with the latest BASS version.

b) "Lound hearable click"
-> I never noticed something like this - in all my tests there is absolutly no click. So I guess this is not related to enc_aacPlus.exe - but might be related to something else in your environment.
Do you have some code on how you use it?

[Mickael]

I am using as a basic source of example to mp3 and ogg. Let me a hint to make use of encryption to MP4 (aacp). Does anyone know what I say a command line?

///VEJA >> SEE

  // setup encoder command-line (raw PCM data to avoid length limit)
  if chk_Mp3.Checked then
  begin
    Content := BASS_ENCODE_TYPE_MP3;
    Comline := PChar(Format('lame -r -x -s 44100 -b %d -', [bitrate]));
  end
  else
  begin
 Content := BASS_ENCODE_TYPE_OGG;
     Comline := PChar(Format('oggenc -r -R 44100 -b %d -m %d -', [bitrate, 16]));
      // for better Quality on ogg just use -q (-2.0 - 10.0}

  end;

I tried :

  else if chk_AACP.Checked then
  begin
    Content := BASS_ENCODE_TYPE_AAC;
    Comline := PChar(Format('enc_aacPlus - - --cbr %d --lc --silent --rawpcm 44100 2 16', [bitrate*1000]));
  end

Result:

---------------------------
Info
---------------------------
The encoder died!
---------------------------
OK   
---------------------------


 

[radio42]

Have you copied the required Winamp files as well?
These are: "enc_aacplus.dll" and "nscrt.dll" from a Winamp 5.2 installation.

[Mickael]

Whooops.. no! It's ok :-)
Thank you!

[Mickael]

and MP2 ? (Mpeg 1 layer 2)

  else if chk_MP2.Checked then
  begin
    Content := BASS_ENCODE_TYPE_MP3;
    Comline := PChar(Format('twolame -r -x -s 44100 -b %d -', [bitrate]));
  end

---------------------------
Info
---------------------------
The encoder died!
---------------------------
OK   
---------------------------

towlame.exe config it's ok  ?

TwoLAME version TwoLame 0.3.12b (http://www.twolame.org)
MPEG Audio Layer II (MP2) encoder
Usage:
        twolame [options] <infile> [outfile]

Both input and output filenames can be set to - to use stdin/stdout.
  <infile>       input sound file (any format supported by libsndfile)
  <outfile>      output bit stream of encoded audio

Input Options
        -r, --raw-input          input is raw signed PCM audio
        -x, --byte-swap          force byte-swapping of input
        -s, --samplerate srate   sampling frequency of raw input (Hz)
            --samplesize bits    size of raw input samples in bits (default 16-b
it)
            --samplefloat        32-bit float raw input
        -N, --channels nch       number of channels in raw input
        -g, --swap-channels      swap channels of input file
            --scale value        scale input (multiply PCM data)
            --scale-l value      scale channel 0 (left) input
            --scale-r value      scale channel 1 (right) input

Output Options
        -m, --mode mode          (s)tereo, (j)oint, (d)ual, (m)ono or (a)uto
        -a, --downmix            downmix from stereo to mono file for mono encod
ing
        -b, --bitrate br         total bitrate in kbps (default 192 for 44.1kHz)

        -P, --psyc-mode psyc     psychoacoustic model -1 to 4 (default 3)
        -v, --vbr                enable VBR mode
        -V, --vbr-level lev      enable VBR and set VBR level -50 to 50 (default
 5)
        -B, --max-bitrate rate   set the upper bitrate when in VBR mode
        -l, --ath lev            ATH level (default 0.0)
        -q, --quick num          only calculate psy model every num frames
        -S, --single-frame       only encode a single frame of MPEG Audio

Miscellaneous Options
        -c, --copyright          mark as copyright
        -o, --non-original       mark as non-original
            --original           mark as original (default)
        -p, --protect            enable CRC error protection
        -d, --padding            force padding bit/frame on
        -R, --reserve-bits num   set number of reserved bits in each frame
        -e, --deemphasis emp     de-emphasis n/5/c (default: (n)one)
        -E, --energy             turn on energy level extensions

Verbosity Options
        -t, --talkativity num    talkativity 0-10 (default is 2)
            --quiet              same as --talkativity=0
            --brief              same as --talkativity=1
            --verbose            same as --talkativity=4


Allowable bitrates for 32, 44.1 and 48kHz sample input (MPEG-1)
  32,  48,  56,  64,  80,  96, 112, 128, 160, 192, 224, 256, 320, 384

Allowable bitrates for 16, 22.05 and 24kHz sample input (MPEG-2)
   8,  16,  24,  32,  40,  48,  56,  64,  80,  96, 112, 128, 144, 160

[radio42]

You should ommit the "-x" flag, as twolame already uses little-endian.
Also you need to either specify the "--samplesize" or the "--samplefloat" flag depending on your input format as well as the "-N" flag.
E.g.:
"'twolame -r -s 44100 --samplesize 16 -N 2 -b..."

[Mickael]

Hello radio42,

  else if chk_MP2.Checked then
  begin
    Content := BASS_ENCODE_TYPE_MP3;
    Comline := PChar(Format('twolame -r -s 44100 --samplesize 16 -N 2 -b %d -', [bitrate]));
  end

---------------------------
Info
---------------------------
The encoder died!
---------------------------
OK   
---------------------------


 

[Mickael]

I'm tested the config on command line:

C:\>twolame -r -s 44100 --samplesize 16 -N 2 -b 128 -
Missing output filename.
TwoLAME version TwoLame 0.3.12b (http://www.twolame.org)
MPEG Audio Layer II (MP2) encoder

Usage: twolame [options] <infile> [outfile]

Try "twolame --help" for more information.

Missing output filename ?

[radio42]

Yes sure, that's why I placed the "..." in my above comment, as I thought you know the rest of the command-line, as it it listed in the twolame help ;-)

[Mickael]

   
Hello radio42.
Sorry I did not understand very well your answer. (I'm French)

This line does not work:

Comline := PChar(Format('twolame -r -s 44100 --samplesize 16 -N 2 -b %d -', [bitrate]));

Do you have any idea why?
Thank you

[Mickael]

Do you have any idea why? 

[radio42]

Yes, you also need to specify the input and output filename as well, that's missing in your command line as far as I can see.

[hwahrmann]

Bernd,

i simply can't make it work.
using the program below doesn't produce any output. Also tried without an Encoding Proc and specified the file name, with the same result.
commenting the enc_aacplus line and use lame, i get a perfect rip of the first track of the CD.
it's just that enc_aacplus does nothing. (Of course, i have enc_aacplus.dll and nscrt.dll in my executable folder).

Also, one other question. how does that work with the --mp4box switch? Can this switch be used together with stdout and an encoing proc?

thanks,

Helmut

using System;
using System.Runtime.InteropServices;
using Un4seen.Bass;
using Un4seen.Bass.AddOn.Enc;
using Un4seen.Bass.AddOn.Cd;

namespace EncTest
{
  class Program
  {
    private static ENCODEPROC _encoderProc;
    private static string _outFile;
    private static System.IO.FileStream _fs = null;
    private static byte[] _data = new byte[1048510]; // 1MB buffer

    static void Main(string[] args)
    {
      Bass.BASS_Init(-1, 44100, BASSInit.BASS_DEVICE_DEFAULT, IntPtr.Zero, null);

      int stream = BassCd.BASS_CD_StreamCreate(0, 0, BASSFlag.BASS_STREAM_DECODE | BASSFlag.BASS_SAMPLE_FLOAT | BASSFlag.BASS_DEFAULT);

      _outFile = "Track1.aac";
      string encoderSettings = "enc_aacplus.exe - - --cbr 128000";
      //string encoderSettings = "lame --preset standard - -";
      _encoderProc = new ENCODEPROC(EncoderSave);
      int encoderHandle = BassEnc.BASS_Encode_Start(stream, encoderSettings, BASSEncode.BASS_ENCODE_DEFAULT, _encoderProc, IntPtr.Zero);

      byte[] encBuffer = new byte[1048510]; // our encoding buffer
      while (Bass.BASS_ChannelIsActive(stream) == BASSActive.BASS_ACTIVE_PLAYING)
      {
        // getting sample data will automatically feed the encoder
        int len = Bass.BASS_ChannelGetData(stream, encBuffer, encBuffer.Length);
      }

      BassEnc.BASS_Encode_Stop(stream);
      Bass.BASS_StreamFree(stream);


      if (_fs != null)
      {
        _fs.Flush();
        _fs.Close();
      }
    }

    private static void EncoderSave(int handle, int channel, IntPtr buffer, int length, IntPtr user)
    {
      if (_fs == null)
      {
        _fs = System.IO.File.OpenWrite(_outFile);
      }
      if (buffer != IntPtr.Zero)
      {
        Marshal.Copy(buffer, _data, 0, length);
        _fs.Write(_data, 0, length);
      }
    }

  }
}

[radio42]

Hi,

as you are using BASS.NET, you might try to use the BASS.NET Encoder framework in the Misc namespace.
Use the "EncoderWinampAACplus", set all your AACplus properties...and then you can use the "EncoderCommandLine" property to see what is different between your command-line and the one generated by the BASS.NET class.

[hwahrmann]

thanks for the quick response.

checked with the EncoderWinampAACplus.
it produced "enc_aacplus.exe - - --cbr 128000 --silent --rawpcm 44100 2 16"

so it was the missing rawpcm spec, which causes the error.

1 things i've noticed though:

 using the command line above with BassEnc results in a file which is exactly 2 times as big as when using it via EncoderWinampAACplus, which shows the same commandline.

any idea?

[radio42]

That sounds strange.
Are both files playing out exactly the same result - even if they are so different in size?

Maybe in one case you are using float channels and in the other 16-bit channels?

[hwahrmann]

yeah it is the BASS_ENCODE_DEFAULT. When setting it to BASS_ENCODE_FP_16BIT, they're the same size.

thanks for your help.

[radio42]

Hi,

a new verison of the enc_aacPlus command-line encoder is available!

Here is the download link: http://www.un4seen.com/filez/4/enc_aacPlus.zip

This new version is based on the aacPlusCLI project as published here: http://aacpluscli.codeplex.com/
* AACPlus v2 Encoder (using Winamp 5.33 or above enc_aacplus.dll and nscrt.dll)
* Coding Technologies encoder 8.0.3
* Build (9349): Mar 18 2008 at 09:00 AM

[Bert B.]

Just a question: Why would you prefer this encoder over Nero's neroAacEnc.exe?
As far as I know this encoder is free, and you need just this executable (no installation of other software).

Bert

[radio42]

That is correct.
The only reason to not use neroAacEnc.exe is, if you want to broadcast AACplus content (e.g. use it with BASS_Encode_CastInit) - as neroAacEnc.exe doesn't support STDOUT (which is needed for broadcasting).
And this was the only reason for making enc_aacPlus.exe - to support STDOUT!

[Bert B.]

Thanks for the explanation.

Regards,
Bert

[3delite]

Can I use enc_aacPlus.exe bundled with my application?

Best regards
3delite

[radio42]

Yes, you can bundle enc_aacPlus.exe with your app...from my point.
But note, that using any AAC decoding or encoding stuff in your app means that your app needs a AAC license from VIA ;-)

[3delite]

I see...

Thanx!

Best regards
3delite

[fmcoder]

Hi everyone.

There seems to be a problem with enc_aacplus.exe (or enc_aacplus.dll).

If I use command line:
enc_aacplus.exe - - --lc --silent --br 64000 --rawpcm 44100 2 16

The output stream is in 32000Hz, but I need 44100Hz. Why it resamples and how can I avoid/control it?

[radio42]

Yes, you are right.
But it seems, that this is a limitation of the enc_aacPlus.dll being used (Winmap).
You might also try the 'official' enc_aacPlus command-line tool: http://aacpluscli.codeplex.com
This gives the exact same results.
I also figured, that other combinations might even fail totally (the resulting aac file is corrupt).
E.g. using a bitrate of 128000 with --lc fails.
But using a bitrate of 32000 with --lc works fine again.

So I am sure, that this looks definitly like a bug in enc_aacPlus.dll - as the .exe simply invokes the winamp .dll encode method with the appropriate values directly as given from the command-line.
So there is absolutly no extra pre- or post-processing involved with regards to the encoding.

However, when you just want to encode LC-AAC I anyhow would recommend to use the FAAC encoder instead of enc_aacPlus.
enc_aacPlus should to my mind only be used in case of HE+-AAC encodeing (aacPlus) - as this is the only format which is not supported by FAAC.

[fmcoder]

Thanks for answer! I'll try to play with faac.exe, because of many complains from users about those 32kHz.

[fmcoder]

Tried to use the faac.exe but no luck.
The command line is:
faac.exe -P -R 44100 -B 16 -C 2 -X -b %d -o - -

But it doesn't produce correct output... does anyone have an example of working with faac?

Also, FAAC seems not support CBR. Is it OK to broadcast in ABR?

Thanks.

[radio42]

When you want to stream (broadcast) AAC, then you must use enc_aacPlus.
The reason is, that SHOUTcast only accepts HE-AAC streams and FAAC only supports LC-AAC (as previously said).

So if you want to use enc_aacPlus, you must also use "--high" option, as this is the only codec which is supported by SHOUTcast.

[fmcoder]

Yes, for HE-AAC (so called AAC+) I use the bass_encaac. I also want to include support for LC-AAC in case some users may need it. So, the problem may be that I use shoutcast, thanks. I didn't know it can't support lc-aac

[fmcoder]

Tried with Icecast and LC-AAC and it still doesn't work. The LC-AAC format is not for broadcasting at all? And, anywhere where I see the AAC it is actually AAC+?

[radio42]

Yes, LC-AAC is not for broadcasting!
Icecast doesn't even support HE-AAC (AAC+) at all - as far as I know.

[fmcoder]

As I tested, AAC+ worked with Icecast.
I was confused, because some users asked for AAC support in my app (when I already had AAC+)

[jeroenp]

You might also try the 'official' enc_aacPlus command-line tool: http://aacpluscli.codeplex.com

Is your current version the same as the one on codeplex?
If not: when should we use which one?

--jeroen

[radio42]

The codeplex one isn't able to to send the encoded data to STDOUT - which is needed for streaming the data to a server.
That was the whole point in doing a fork here.

[jeroenp]

The codeplex one isn't able to to send the encoded data to STDOUT - which is needed for streaming the data to a server.
That was the whole point in doing a fork here.

Basically that was my underlying question; since you pointed to the original, I wondered if it too supported STDOUT as encoding target.
Good to hear you confirmed only your version does STDOUT.

--jeroen

[jeroenp]

Is this the one that includes your patches?
https://github.com/tmkk/aacPlusCLI
--jeroen

[radio42]

No, I guess they never integrated my patches into the original code!

[jeroenp]

Would you willing to provide your patches for integration?
Thanks,
--jeroen

[radio42]

They should all be included in the link of the first post.
What is missing, or not working?

[jeroenp]

(Sound of head smashing table): I missed that link, had only see the binary before. Sorry (: