BeHappy - AviSynth based audio transcoding tool (UPD 19-07-2006)

[tebasuna51]

Originally Posted by Menedas

Of which DSP do you talk now? Not Timestretch?

Yes TimeStretch. There are 3 modes in 'Rate Control':
- Rate ...
- Pitch ...
- Tempo ...

The Rate mode don't need sophisticated algorithms to do the job, only change the samplerate then the waveform is very similar and only the pitch change can be detected. This is the -slowdown/speedup from eac3to.

Pitch or Tempo modes need more complex job.
You can try also different parameters in TimeStretch AviSynth function

[lchiu7]

Our new DVB-T broadcasts are using this audio format. About the only way to hear the recorded files is to have PowerDVD8 which has a DS filter that can handle this audio format.

Since Behappy uses DS I wonder if it's possible to use this tool to convert this audio format to something common - say MP4 or even 2 channel AC-3 so that it could be muxed back with the original (or even re-compressed) AVC video into something that is easier to play.

Thanks

[Menedas]

Ok, then I have understood you right. But how can you explain, that there is the same effect hearable with the Rate option? Hence, the effect I can hear could not come from the pitch. Thats what I wanted to say the whole time.

I wasn't really aware that the speedup option of eac3to is maybe the thing I wanted. I tested it and it does not have the same "bug" as the AviSynth function (Rate). Thank you for that information. But thats only a solution for eac3 files. I need also something for AC3 or DTS files.

[tebasuna51]

Originally Posted by lchiu7

Our new DVB-T broadcasts are using this audio format. About the only way to hear the recorded files is to have PowerDVD8 which has a DS filter that can handle this audio format.

Since Behappy uses DS I wonder if it's possible to use this tool to convert this audio format to something common - say MP4 or even 2 channel AC-3 so that it could be muxed back with the original (or even re-compressed) AVC video into something that is easier to play.

Seems there are also a free LATM aac DS decoder

You can try open the file with the DirectShowSource method.

[tebasuna51]

Originally Posted by Menedas

Ok, then I have understood you right. But how can you explain, that there is the same effect hearable with the Rate option? Hence, the effect I can hear could not come from the pitch. Thats what I wanted to say the whole time.

I can't explain a effect that I don't hear.

I wasn't really aware that the speedup option of eac3to is maybe the thing I wanted. I tested it and it does not have the same "bug" as the AviSynth function (Rate). Thank you for that information. But thats only a solution for eac3 files. I need also something for AC3 or DTS files.

I'm happy you found the solution because eac3to also decode ac3 and dts files.

[Menedas]

Originally Posted by tebasuna51

I can't explain a effect that I don't hear.

Strange. You can't even hear it with headphones?

[lchiu7]

Originally Posted by tebasuna51

Seems there are also a free LATM aac DS decoder

You can try open the file with the DirectShowSource method.

Tried that (under Vista SP1). Demuxed the audio from the ts file using tsremuxer (saved as a aac file)

Had the Monogram filter already installed

Opened the aac file using behappy and chose Directshow (also tried avisynth).

Both times in a few second Behappy just died (program has stopped responding).

So it looks like Behappy is unable to open these files

[gtpboy]

Maybe I’m doing something wrong but I'm trying to take a 6 channel AC3 file splitting into individual mono WAV files then re encode it into a single WAV file in WME.

Well once BeHappy gets done splitting the AC3 file into the individual WAV files I go to convert it in WME and it tells me that the source files need to be mono WAV files, well I thought that’s what I just did. I've double and triple checked all the settings in BeHappy but can't figure it out.

Is there an easier way to do this like a straight encode AC3->WMA 10 instead of splitting and then rejoining?

[tebasuna51]

Originally Posted by gtpboy

Maybe I’m doing something wrong but I'm trying to take a 6 channel AC3 file splitting into individual mono WAV files then re encode it into a single WAV file in WME.

What is WME?
Do you need a single WAV or a WMA?

Well once BeHappy gets done splitting the AC3 file into the individual WAV files I go to convert it in WME and it tells me that the source files need to be mono WAV files, well I thought that’s what I just did. I've double and triple checked all the settings in BeHappy but can't figure it out.

BeHappy can decode ac3 files to a single WAV multichannel (Destination Wav Writer) file or to monowav files (Destination WavSplit @ Mono Wav's).

Is there an easier way to do this like a straight encode AC3->WMA 10 instead of splitting and then rejoining?

If MS supply a WMA encoder with STDIN input we can add a direct transcode AC3->WMA in BeHappy.

[Snowknight26]

WME = Windows Media Encoder.

[tebasuna51]

Originally Posted by Snowknight26

WME = Windows Media Encoder.

Sorry, I don't know this tool.
I supose you want output a multichannel WMA.

And WME need 6 monowavs always to encode a multichannel WMA?
And WME don't recognize the monowavs generated by BeHappy-WavSplit?

[gtpboy]

BeHappy can decode ac3 files to a single WAV multichannel (Destination Wav Writer) file or to monowav files (Destination WavSplit @ Mono Wav's).

I did see that option in there but didn't know it generated a multichannel WAV file I'll have to try that thanks.

And WME don't recognize the monowavs generated by BeHappy-WavSplit?

Aparently i've tried it on a few different AC3 files and get the same message "source must be mono WAV file"

Oh well as long as the other method works i'll be fine Thanks again

WME is a fickle program some AC3 files it will transcode with no problems others it will generate a "source file type is invalid" error same with some DTS files

[DiGiT@LON€]

Hi everyone.
I'm an Italian user, sorry for my bad English.

I have to make 2 questions:

- I have an ac3 2.0 audio file. I want to convert it in wav with behappy latest release.
I select wav writer like encoder, and the encode works good.
But the output wav file don't have sound.
I can see from foobar that it is a PCM 32 bit floating point, but there's not sound.
How can I convert better?

- Does Behappy add a delay to output file? If yes, how much?

Thanks...

[tebasuna51]

Originally Posted by DiGiT@LON€

- I have an ac3 2.0 audio file. I want to convert it in wav with behappy latest release.
I select wav writer like encoder, and the encode works good.
But the output wav file don't have sound.
I can see from foobar that it is a PCM 32 bit floating point, but there's not sound.
How can I convert better?

This is a NicAudio ac3 decoder behaviour more than BeHappy related.

When the decoder found a valid ac3 frame set some basic parameters (num_channels, samplerate, bitrate) and after reject (filling with silence) any other frame don't match the initial basic parameters.

Probably your ac3 source is from a TV capture and you have some initial frames 2.0 (commercials) and after change to 5.1 (movie), sorry but NicAudio can't begin supply 2.0 and change to 5.1 on the fly. You can use DelayCut to check the ac3 file and cut the initial 2.0 frames (if is the problem).

- Does Behappy add a delay to output file? If yes, how much?

Behappy have a box to include any desired delay. By default BeHappy don't add delay.

NicAudio.dll v2.0.2 ac3 decoder can add delay (a multiple of 32 ms) when found invalid data (until 1 MB) before the first valid ac3 frame, then can compensate pseudo-delays in VirtualDub style. If you don't want this delay you can use DelayCut to fix the ac3 before decode.

There are also little delays introduced by encoders, for instance ac3 encoders do a 5.333 ms delay (with Aften you can disable this delay with the -pad 0 parameter)

[DiGiT@LON€]

Originally Posted by tebasuna51

This is a NicAudio ac3 decoder behaviour more than BeHappy related.

When the decoder found a valid ac3 frame set some basic parameters (num_channels, samplerate, bitrate) and after reject (filling with silence) any other frame don't match the initial basic parameters.

Probably your ac3 source is from a TV capture and you have some initial frames 2.0 (commercials) and after change to 5.1 (movie), sorry but NicAudio can't begin supply 2.0 and change to 5.1 on the fly. You can use DelayCut to check the ac3 file and cut the initial 2.0 frames (if is the problem).

Yes, you're right. It comes from a TV capture.
Can you suggest me another tool that convert ac3 (in this status) in wav?
Have I to use Delaycut imperatively?

Originally Posted by tebasuna51

There are also little delays introduced by encoders, for instance ac3 encoders do a 5.333 ms delay (with Aften you can disable this delay with the -pad 0 parameter)

Does Ac3 encoder in BeHappy have this behaviour?
If yes, have I to set -5.333 in BeHappy for synchronizing?

[DiGiT@LON€]

I have resolved with Azid-BeSweet.
Now I want to know only if azid is between those encoders that apply a delay during the encode.

Then, if someone explains me why the encode works with azid-beesweet, I appreciate...
Thanks...

[Kurtnoise]

@Tebasuna:

I found a bug in the AvisynthWrapper. Using your dll + the wrapper with megui, I got a buffer overrun issue.

Here is the fix...

[tebasuna51]

You are rigth Kurtnoise. Next release must correct that. Thanks.

Isn't a problem for BeHappy because seems never read video frames but MeGUI ...

I don't know if AvisynthWrapper.cs can work with MeGUI my unique change was 'dimzon_avs_init' -> 'dimzon_avs_init_2'

[Kurtnoise]

well...the AvisynthWrapper.cs is the same (quite normal because we uses the same lib ). The main difference comes from the encoder routines (wav header writing, etc...). Yesterday, I tried to drop the ConvertAudio16Bits() restriction from the megui script but unfortunately, it produces some garbage as ouput whereas with BeHappy and the same script, all it's fine. So, I suspect something wrong with the wav header. I've seen that the wav header writing is more accurate with BeHappy (wav > 4GB detection, different headertypes, etc...) but I'm busy with other things right now so I can't check it out more carefully...

[tebasuna51]

You are writing always INT wav files.

In AviSynthAudioEncoder.cs (line 482) you need only write the correct Format_tag:

- target.Write(BitConverter.GetBytes((short)0x01), 0, 2);
+ target.Write(BitConverter.GetBytes((a.SampleType==AudioSampleType.FLOAT) ? (short)0x03) : (short)0x01), 0, 2);

Where (in AviSynthWrapper.cs):

	public enum AudioSampleType:int	{
		Unknown=0,
		INT8  = 1,
		INT16 = 2,
		INT24 = 4,
		INT32 = 8,
		FLOAT = 16
	};