how to convert mp2 audio to lpcm?

[gareth41]

Ive got a 19Gb raw TS transport stream from the hdd of my sat receiver, h264 video 1920x1080i and MP2 audio 256kbps. I want to burn this to bluray which would be easy enough if the audio was AC3, but its MP2 which is not a supported bluray audio format (correct me if im wrong) so I need to somehow convert this to LPCM then remux to m2ts.

So far I have gotten as far as converting the MP2 audio to a PCM wav 48000Hz 16bit, but this is not LPCM. Could anyone advice of a method/program to convert the MP2 audio to LPCM?

[gareth41]

Dont worry, have decided just to convert the mp2 to ac3 instead. Fortunately most of the HD channels I receive contain an AC3 audio track so its only a matter of feeding into txmuxer. Its only a select few channels which only have mp2 audio and need converting.

[loster]

besweet or eac3to

[El Heggunte]

To whom this may interest:

[C:\]
=>wav2lpcm

wav2lpcm - Copyright (C) 2005 Dave Chapman
Latest version available from http://dvd-audio.sourceforge.net/

This is free software; see the source for copying conditions. There is NO
warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.

USAGE: wav2lpcm input.wav > output.lpcm

wav2lpcm converts a WAV file to LPCM suitable for multiplexing into a
DVD-Video compliant MPEG-2 program stream using the mjpegtools
"mplex" utility.

The LPCM data is written to stdout.

[C:\]
=>fixpcm

################################################## ###########
Pcm2Tsmu-C v1.0.0.1: Fix pcm for TsMuxer format. (3r1c 2009)
Based on Pcm2Tsmu v1.3 by Tebasuna. Rewritten in C language.
For distribution with mkv2vob. http://www.mkv2vob.com/
################################################## ###########

Usage:
fixpcm <input.pcm> [output.pcm] [-i #] [-c #] [-s #]

Where:
<input.pcm> must be a lpcm up to 8 channels, int 16-20-24 bits,
and 48, 96 or 192 KHz.

[output.pcm] If not present, input.pcm is used suffixed with: _tsm

[-i #] BitsPerSample Integer. Default 24 (20 not tested).

[-c #] NumChannels. Default 6.

[-s #] SampleRate. Default 48000.

[-le] Input is Little Endian.

Note: To use STDIN/STDOUT use - as filename.

[C:\]
=>

[jman98]

So far I have gotten as far as converting the MP2 audio to a PCM wav 48000Hz 16bit, but this is not LPCM. Could anyone advice of a method/program to convert the MP2 audio to LPCM?

Cool Edit is the only thing I know of that has a crazy way to generate a WAV file that somehow is not kosher for LPCM purposes and you have to work at it to get it to do that. BeSweet would be my suggestion, but if you found a way to convert it to AC3 that's good.

You are right that BluRay does not accept MP2 audio. There's actually nothing wrong with MP2 as from a quality perspective 256 Kbps MP2 audio is really quite excellent quality audio. Philips apparently threw some kind of hissy fit in the DVD standards days and demanded that MP2 get placed in, so they added to the PAL standard and left it out of the NTSC one. Philips seems to have basically given up on MP2 for the most part. It's lack of realistic support for multi-channel audio (you can theoretically produce MP2 multi-channel audio, but I say it's not "realisitic" because almost no playback devices anywhere support it and it's darn near impossible to find encoders that can create it) pretty much doomed it as it has no advantages that LPCM or AC3 can't offer and both are better supported.

[Ennio]

... I have gotten as far as converting the MP2 audio to a PCM wav 48000Hz 16bit, but this is not LPCM...

Are you sure it's not lpcm? wav files generally are lpcm; if the quantization of pcm is linear, it is - as it says - lpcm (the word pcm is often meant as lpcm).

http://en.wikipedia.org/wiki/Linear_PCM

Please correct me if I'm wrong or miss-interprete thangs? Much appreciated.

cheers

Ennio

[El Heggunte]

In general,

"LPCM" means "(.WAV minus WAV header) ++ reverse byte-order"

[Cornucopia]

Yeah, usually, WAV LPCM uses Intel-ordered (LSB or "Little Endian") samples, whereas RAW LPCM that is used as input for Muxing with MPEG2 video (PS->DVD, TS->BD, etc) is usually Motorola-ordered (MSB or "Big Endian") samples (depends on the app). Simple enough to swap the byte ordering.

The maker of AC3Filter tool also makes a byte-order swapping CLI tool called, "swab".
Also, DVDLab has a tool in it's menu that can swap bytes between LPCM <-->WAV.
And, IIRC, CoolEdit (probably Audition, too, maybe soundforge) can choose which byte ordering when exporting to raw LPCM.

Scott

[Ennio]

Ok, this is new to me. Thanks for the head-up, in my case. Any good explanatory site suggestions - besides search engines - where I can read about these header thangs?

Thanks

@gareth41: have you looked into this? https://forum.videohelp.com/threads/236976-wav2lpcm-for-windows

cheers

Ennio

[Cornucopia]

AFA header info: look up using combinations of "WAV" "AVI" "RIFF" "Riff view" "Container" "header" and you should get to some sites that describe how those filetypes encapsulate the media streams (though most of those sites will be quite tech-oriented, full of jargon).

Most multimedia containers/filetypes are built generally like this:

Key: H=Header, F=Footer, SH=SubHeader, M1=MediaStream1, M2=MediaStream2

H_SH_M1a_M2a_SH_M1b_M2b_SH_M1c_M2c_SH_M1d_M2d_SH_M1e_M2e_...SH_M1z_M2z_F

This is called "interleaved" & "packetized". It's possible for it to be less interleaved:

H_SH_M1a_SH_M1b_SH_M1c_SH_M1d_...SH_M1z_SH_M2a_SH_M2b_SH_M2c_SH_M2d_...SH_M2z_F

Or less packetized:

H_SH_M1a_M1b_M1c_M1d_M1e_...M1z_SH_M2a_M2b_M2c_M2d_M2e_...M2z_F

A "minimalistic" container might look like this:

H_M1

Scott