eac3to - audio conversion tool

[tebasuna51]

command line: eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.pcm
...
[a03] Creating file "C:\movie\HIC.pcm"...
...

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\HIC.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\HICUPDATE.pcm"

How do you convert HIC.pcm to HICUPDATE.pcm?

Try with:
eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.w64

TsMuxer 1.10.6 accept w64 like PCM container.

[dcmo]

How do you convert HIC.pcm to HICUPDATE.pcm?

Try with:
eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.w64

TsMuxer 1.10.6 accept w64 like PCM container.

I've been converting with pcm2tsmu. My question now is is
.w64 going to be equivalant to DTS HD MA/PCM or is it a lower grade sound quality. I'm really not interested in a 2.0 solution or any loss in sound quality.

[nurbs]

It's going to be exactly the same as the source.

[tebasuna51]

I've been converting with pcm2tsmu.
Please put command line
My question now is is .w64 going to be equivalant to DTS HD MA/PCM or is it a lower grade sound quality.
PCM in WAV or W64 container and raw LPCM have absolutely the same uncompressed audio data. You can verify the sizes, only differents by a few bytes for the header.

[dcmo]

Been pretty busy working today, just realized what you meant by command line. I just assumed it was pretty universal on pcm2tsmu input. Here is what was input:
pcm2tsmu.exe C:\movie\hic.pcm C:\movie\hicupdated.pcm -i 24 -c 8 -s 48000

Getting ready to give w64 a try now, hope it comes thru for me.

[arty]

D:\util\eac3to>eac3to i:\temp\movie.mkv 3:d:\trans.flac
MKV, 1 video track, 2 audio tracks, 2:30:25, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
"movie h264 stream"
2: DTS, English, 5.1 channels, 24 bits, 1509kbps, 48khz
"dts core track"
3: FLAC, English, 5.1 channels, 2:30:25, 24 bits, 48khz
"FLAC created by eac3to3.17 + Sonic 4.3.0 filters"
a03 Extracting audio track number 3...
a03 Decoding FLAC...
a03 Encoding FLAC with libFlac...
a03 Creating file "d:\trans.flac"...
----------------------------------

is this "demux" method lossless? couldn't find any other way to demux a flac from mkv - mktoolnix creates flac inside ogg :(

[Snowknight26]

is this "demux" method lossless?
Of course it is, you're going from lossless -> lossless.

couldn't find any other way to demux a flac from mkv - mktoolnix creates flac inside ogg :(
You didn't look hard enough then. There's a switch to have it just write a raw FLAC track without doing an OggFLAC.

[arty]

Of course it is, you're going from lossless -> lossless.


You didn't look hard enough then. There's a switch to have it just write a raw FLAC track without doing an OggFLAC.

thank you!

ahh, the --no-ogg param, so true :)

[dcmo]

.w64 was a complete bust for me, TS Muxer locked up on two different movies on me (one with TrueHD and the other with DTS HD MA). Both times with this:

LPCM bad frame detected. Resync stream.

Settings for TS Muxer:

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\ppl.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\ppl.w64"

SmartLabs tsMuxeR. Version 1.10.6 http://www.smlabs.net
Decoding H264 stream (track 1): Profile: High@4.1 Resolution: 1920:1080p Frame rate: 23.976
H.264 stream does not contain fps field. Muxing fps=23.976
Decoding LPCM stream (track 2): Bitrate: 6912Kbps Sample Rate: 48KHz Channels: 5.1 Bits per sample: 24bit

Actually though it wasn't a complete bust, it did about 35% of the THD version and about 25% of the DTS before it locked up on me. Both played fine on the computer, haven't checked it out on the PS3 as of yet.

Okay, played it on the PS3 and it came out great for what was there, does anyone know what caused TS Muxer to stop muxing. Or rather why a bad frame was detected. How do I solve this problem.

On what was there, there didn't appear to be any mapping problems (I really like that). The sound sounded pretty much identical, I think I would need two audio systems to be able to check it out.
However the tv upstairs still didn't play it properly, it was like the center channel (main dialog) was missing although on the system downstairs it came thru properly. It's an older receiver (5 years maybe, no hdmi, no advanced audio). Should I swap cables around or do I just need a new receiver.

Thanks to all for you help!

[tebasuna51]

...
LPCM bad frame detected. Resync stream.
...

Seems a TsMuxer problem. LPCM (or WAV or W64) don't have frames, only audio data. The audio data can be correct (music, voices, ...) or incorrect (noise, wrong channels, ...) but don't exist a method to detect a bad frame.

[mikelebron]

Guys.. Is it possible for me to remux my MKVs that have FLAC for audio and change the AUDIO to LPCM? Reason why I am asking is I want to stream my MKVs to the PS3... Am I going about it the wrong way?

[Inspector.Gadget]

Mikelebron: When you create the Matroska file from the original Blu-ray, just output PCM (WAV) audio instead of FLAC. Nothing requires you to use FLAC when demuxing a disc with eac3to...

[dcmo]

Seems a TsMuxer problem. LPCM (or WAV or W64) don't have frames, only audio data. The audio data can be correct (music, voices, ...) or incorrect (noise, wrong channels, ...) but don't exist a method to detect a bad frame.

So I should take this to the TS Muxer section? I saw this problem described over there (in April), but there never was really an answer for it.

I was reading in the thread about first converting it to WAV, then to W64; do you think this would help.

If so how does one go about converting it from wav to w64; I know that eac3to.exe E: 1) 2: C:\movie\movie.h264 3: C:\movie\movie.wav would get it to a wav file (although I've read that wav has problems with over 4G files), but how do you then convert that wav file to a w64 file. I assume it would be a command line, what would that line look like?

I'm going to try to convert it to pcm tonight and see if TS Muxer still has problems with muxing it.

[tebasuna51]

...
I'm going to try to convert it to pcm tonight and see if TS Muxer still has problems with muxing it.
TsMuxer don't accept standard LPCM (.pcm from eac3o), for that I wrote Pcm2tsmu.

TsMuxer always accept .wav files but only <4GB
Last versions of TsMuxer say accept .w64 files (same data than .wav but with a header to support >4GB).

I don't know how solve your problem.
My recommendation is use AC3 640 Kb/s, only with very good audio equipment in a conditioned room you can listen some difference.

[dcmo]

I was going to use pcm2tsmu just to see if it was w64 that was causing the problems and not the discs. I'm sure it's w64, I just wanted to be sure.

After reading some more, what about splitting the file and rejoining them later (that way I could do it in a wav file).

I just can't downgrade the sound; if the choice is between streaming with downgraded sound or not streaming at all then not streaming at all is hands down the winner (it's really not even an option).

I will just go the re-build route and burn it to a physical disc. Anyway can I split them and rejoin them, would I be able to tell where it was happening when I watched it.

Thanks again.

[utenteanonimo64]

I have just discovered eac3to and I am really impressed. In a matter of days it has become a central part of my audio processing workflow. There is only one thing I am missing (or I haven't been able to find out): I would like an option to have the audio output from a bluray stream split into several files one for each chapter.
Is Madshi still supporting this software and adding new features? Or is there an automated way to achieve this?
I have seen that someone has written a GUI called "HDConcertRipper" for this purpose but it gives me all sorts of errors on most of my discs...

[tebasuna51]

Try a search in this thread with your title "splitting audio by chapter"

For instance: http://forum.doom9.org/showpost.php?p=1209593&postcount=6845

[deebo]

some weird feature with powershell in windows7 (works fine without the -quality switch):

PS D:\> eac3to 00002.m2ts 8:commentary.aac -quality=0.35
...
Track 2 is used for destination file ".35".
This audio conversion is not supported.

works fine with cmd.exe:


D:\>eac3to 00002.m2ts 8:commentary.aac -quality=0.35
...
a08 Extracting audio track number 8...
a08 Patching bitdepth to 24 bits...
a08 Decoding with libav/ffmpeg...
a08 Reducing depth from 64 to 32 bits...
a08 Encoding AAC <0.35> with NeroAacEnc...
---

[utenteanonimo64]

Try a search in this thread with your title "splitting audio by chapter"

For instance: http://forum.doom9.org/showpost.php?p=1209593&postcount=6845

Well I wish Madshi changed his mind.... personally it's the only feature I miss...

[AnryV]

eac3to can't decode 1.0 DTS-HD MA
eac3to v3.17
command line: eac3to fre.dtsma fre.wavs
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

eac3to v3.17
command line: eac3to fre.dtsma fre.wavs -sonic
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 896126544. <ERROR>

[dcmo]

Actually I got it figured out. It wouldn't do a wav or w64 file, but it would do a pcm (with pcm2tsmu) file. After having checked that out, I converted the original audio to a wav and then to w64. Turned out perfect, sounds great. As far as the issue upstairs, picked up a cheap Insignia receiver (HDMI input with HD audio codecs) on sale at Best Buy which solved the sound issue upstairs.

[tebasuna51]

eac3to can't decode 1.0 DTS-HD MA

Please, can you upload a sample?.
3 MB is enough:

eac3to fre.dtsma sample.dtshd -3mb

[Inspector.Gadget]

AnryV, maybe i'm missing something, but could the error be the result of the wavs switch for a mono source. What happens when you output to audio.flac instead?

[AnryV]

Please, can you upload a sample?.
3 MB is enough:

eac3to fre.dtsma sample.dtshd -3mb

http://rapidshare.de/files/48730597/sample_Tonton.dtshd.html

This occurs not only with this sound.

[tebasuna51]

This occurs not only with this sound.

Yes, seems a bug in ArcSoft decoder (same error with output to flac or wav).
I don't have Sonic to test.

BTW, the core bitrate, 768 Kb/s, is more than enough for a mono audio. Must be dificult diference from original.

[AnryV]

Yes, seems a bug in ArcSoft

And Sonic?
eac3to v3.17
command line: eac3to fre.dtsma fre.wavs -sonic
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 896126544. <ERROR>

TMT3 play it normally. I think that this is a bug of eac.

[xkodi]

TMT3 play it normally. I think that this is a bug of eac.

and why you are so sure that TMT3 plays the DTS-HD MA extension rather than DTS core, which i'm sure eac3to can also extract and decode correctly.

[AnryV]

and why you are so sure ...
Not sure... now. :(

[dcmo]

Been trying to get Ice Age 3 done, but Arc Soft is responding with strange set-up. The completed file outputs the center out the right channel, so a couple questions please. When channel remapping which channels do -0,1,2,3,4,5 correspond to? I take this is for 5.1, what happens when you have a 7.1 track? I assume you run the -log to find out what the channels are listed at, is this a correct assumption?

[Thunderbolt8]

And Sonic?


TMT3 play it normally. I think that this is a bug of eac.
afaik its a bug of the arcsoft decoder, we've already had this example that it cannot handle dts-hd ma 1.0 files

[AnryV]

afaik its a bug of the arcsoft decoder, we've already had this example that it cannot handle dts-hd ma 1.0 files
What about Sonic decoder?

[asc28]

Is there any way to recalculate and add back dialog normalization to AC3 tracks, after eac3to has stripped them?

[edit] realized that many studios use an arbitrary figure, so might not be able to recalculate. But how about using a ReplayGain like technique to find the optimal dB to be added and then adding it to an already encoded AC3 track?

Also, I know this has been asked a couple times before, but no one seems to know how exactly to go about reinserting dialnorm metadata without re-encoding.

[Esurnir]

I'm going to sound dumb, but which product from arcsoft still include the DTS-HD filter decoder? Does TotalMedia Theater 3 work ? Any special editions? (platinum?)

[Thunderbolt8]

What about Sonic decoder?
dont know to be frank.

[IanD]

I've been converting a DTS HD MA and DD2.0 soundtrack to wav with Eac3to, using -core and -down2 for the first, in order to align them.

However neither of the resulting wav files are directly readable by Goldwave, Cooledit or CDwave as it seems they are "raw". No matter what bit depth, signature or sample rate I choose, I only get noise. Strangely I am able to play the wav files with Directshow no problem.

How do I make the wav files readable in the sound editor applications?

It's frankly annoying that the output isn't wav standard.

[kypec]

Try to add switch -simple and see if it helps your applications to recognize such stereo WAV files. If yes then you should blame those apps that they don't support standard WAV headers.

[tebasuna51]

GoldWave (from v5.02, 2004? ) read WAVE_FORMAT_EXTENSIBLE headers without problems.

[73ChargerFan]

I've been converting a DTS HD MA and DD2.0 soundtrack to wav with Eac3to, using -core and -down2 for the first, in order to align them.

However neither of the resulting wav files are directly readable
Perhaps the DTS core is being converted to a dts-wav file, instead of being decoded?

eac3to (DTS or DD file) filename.wavs should decode, one channel per file.

[IanD]

Using -simple seems to have done the trick and both converted DTS HD MA and DD2.0 source files are now recognised as stereo wav.

However, I'm experiencing problems trying to synchronise the tracks: the DTS HD MA soundtrack is identified by eac3to as 24fps (not 23.976fps), whilst the commentary track I'm trying to synchronise to it is from a PAL source. Using the switches -25.000 and -ChangeTo24.000 results in the commentary being about 500ms longer than the soundtrack when the beginnings are synchronised.

I thought it might be possible to use the source switch -24.99x to fine tune the commentary conversion, in case the PAL source is not quite 25fps, but eac3to refuses to accept anything other than -25.000

Any thoughts?

Unfortunately I don't have access to an NTSC commentary, only a PAL one. It's quite close at the moment but not close enough because the commentary also has parts of the soundtrack that need to be in-sync with the on-screen action.

[Vick]

Many thanks to Christopher Key for providing the HDCD decoder!

Hi madshi

Whether Chris has given you the HDCD decoder source code?
He does not respond to email (cjk32 cam ac uk) :confused:

Thnx
Vick
p.s. sorry for my poor english

[pcordes]

I'm an old-school GNU/Linux command line junkie. Running eac3to inside
screen (http://en.wikipedia.org/wiki/GNU_Screen), the output comes out like this:


$ alias eac3to='wine /usr/local/.../eac3to.exe'
$ eac3to foo.eac3
fixme:mixer:ALSA_MixerInit No master control found on HDA ATI HDMI, disabling mixer
fixme:reg:GetNativeSystemInfo (0x32fc80) using GetSyE-AC3, 5.1 channels, 0:31:59, 1536kbps, 48khz
# or, discarding wine's messages: eac3to 2>/dev/null
$ eac3to 2(1) Haali Matroska Muxer


I spent a while looking for --help output, until I eventually realized that its output was designed for a Windows command window, not a VT100 terminal emulator. It happens to work outside of screen, in just gnome-terminal, and in native xterm, though.

eac3to outputs a bunch of ^h (ascii 0x08) characters at the start of every line, and maybe under screen, reverse-line-wrap happens, and the cursor backspaces up to the previous line, so every line of text overwrites the one before. If I redirect the output to a file,

$ eac3to > eac3to.txt 2>/dev/null
$ file eac3to.txt
eac3to.txt: ASCII English text, with CRLF line terminators, with overstriking
$ hexdump -C -v eac3to.txt
00000000 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000010 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000020 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000030 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000040 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000050 20 20 20 20 20 20 20 20 20 20 20 20 20 20 0d 0a | ..|
00000060 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000070 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000080 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000090 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000a0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000b0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000000c0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000000d0 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 08 08 | ........|
000000e0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000f0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000100 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000110 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000120 08 08 08 08 08 08 08 08 08 08 65 61 63 33 74 6f |..........eac3to|
00000130 20 76 33 2e 31 37 2c 20 66 72 65 65 77 61 72 65 | v3.17, freeware|
00000140 20 62 79 20 6d 61 64 73 68 69 2e 6e 65 74 20 20 | by madshi.net |
00000150 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000160 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000170 20 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 08 | .......|
00000180 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000190 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001a0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001b0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001c0 08 08 08 08 08 08 08 08 08 08 08 20 20 20 20 20 |........... |
000001d0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000001e0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000001f0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000200 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000210 20 20 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 | ......|
...


But the overstriking doesn't have any effect anywhere: xterm or gnome-terminal. (it visually looks the same as regular text). In VT100, ^h is cursor-movement, not highlighting. If eac3to is doing that itself on purpose, please don't, at least under wine. The CRLF line endings are not a problem.

I'd also love to have an option to not play sound. I haven't tried just renaming the .wav files. Maybe you could add a -unix option, or check an environment variable, to turn off sound and not print funky text. It's harder to deal with options at the end, instead of as the first arguments (e.g. an alias doesn't work), but a wrapper script like

#!/bin/sh
wine /.../eac3to.exe "$@" -unix 2>/dev/null
# or use a wine cmdline arg to filter out the messages it always prints, instead of redirecting to /dev/null

would do the trick.

apparently, wine devs
don't want apps to detect wine (http://forum.winehq.org/viewtopic.php?p=25906&sid=4a6025570ec9cb97aa5e4d952cf6ad88), so the -unix option would be a nice feature. Or have a -nosound option, or a -quiet option. (OTOH, -quiet usually means no messages, not literally quiet.) I'm worried that -nosound would cause confusion, though, since someone might think it means exclude audio tracks. But it's -nosound, not -noaudio...

Anyway, that's my feature request, please and thank you.

[samepaul]

I encounter problem. During THD -> AAC encoding the warning "Clipping detected, a 2nd pass will be necessary." is written and then eac3to indeed performs 2nd pass, making final audio very quiet and practically useless. Is there way to ignore clippings and avoid "2nd pass"?
Also eac3to writes "Reducing depth from 64 to 32 bits". What does it mean? Looks like volume related issue as well... if so, is it possible to prevent this downgrading too?

[tebasuna51]

I encounter problem. During THD -> AAC encoding the warning "Clipping detected, a 2nd pass will be necessary." is written and then eac3to indeed performs 2nd pass, making final audio very quiet and practically useless. Is there way to ignore clippings and avoid "2nd pass"?

You can add the parameter:
-no2ndpass
but is a very little diferent volume.

Also eac3to writes "Reducing depth from 64 to 32 bits". What does it mean? Looks like volume related issue as well... if so, is it possible to prevent this downgrading too?

Is the size of the audio sample, trust in eac3to, NeroAacEnc don't support 64 bit float audio samples.

[samepaul]

You can add the parameter:
-no2ndpass
but is a very little diferent volume.


Thank you! It did the magic - volume remained good. Seems anti-clipping algorithm requires some improvements :)

[tebasuna51]

... Seems anti-clipping algorithm requires some improvements :)

Then there are big volume difference ...
Please, can you upload a sample?

[samepaul]

Seems you're right. I tried to notch the place where clipping occurs, cut it out and encoded with and without 2nd pass.
There was difference in volume, but not so dramatic as it was when whole track was encoded. So I don't think that these samples are good for debugging.

I think it's not completely fault of encoder. I guess the track has places with really hard overload, but when eac3to performs 2nd pass it lowers volume of whole track evenly. Of course, after this previously clipped parts (blows, shootings etc) sound better, but "normal" parts where people talk become indiscernible. If I have tool to detect exact places where eac3to detects maximal clipping I would prepare sample for you. But without this I'd have to submit whole 700mb track , which I don't know how to do.
Btw, I've mistaken when I wrote "TrueHD". It was actually AC3.

But nevertheless, regarding to volume optimization, I think if eac3to performs 2 passes it can normalize levels non-linearly.

[honai]

I think if eac3to performs 2 passes it can normalize levels non-linearly.

... which is called Dynamic Range Compression (DRC) and is ignored by eac3to by default. DRC means you lose fidelity, and that is certainly not desirable if you want to preserve the quality of the original audio track.

[samepaul]

... which is called Dynamic Range Compression (DRC) and is ignored by eac3to by default. DRC means you lose fidelity, and that is certainly not desirable if you want to preserve the quality of the original audio track.

Uhum... and making audio quiet so much that you hadly can distinguish what people say - this is what you call "high fidelity"? If so, I'd stick with normal fidelity ;)

1st of all, DRC does not lower fidelity more than any volume adjustment, which is done by eac3to anyway. And really negligible comparing to quality loss due to audiocompression.
2nd, what I propose is not really DRC. Range compression only performed around overloaded zones, leaving ~90% (depending on content) of audiotrack intact.

[IanD]

The switches -25.000 (source) and -ChangeTo24.000 imply that they are floating point values that "may" accept any floating point value, but in fact seem to be hardcoded to -23.976, -24.000 and -25.000 and nothing else is accepted.

Is there any way to define variable floating point values to finetune these parameters?

Is there an alternate way to perform non-standard resampling and time expansion/compression?

[tebasuna51]

...
Is there any way to define variable floating point values to finetune these parameters?
To make the job eac3to use SSRC libraries to change the audio samplerate.
And not all conversion are possible. You can know available values here: http://avisynth.org/mediawiki/SSRC

Is there an alternate way to perform non-standard resampling and time expansion/compression?
There are commercial software like TimeFactory (http://www.prosoniq.com/editing-products/timefactory-2/)

And also AviSynth methods like TimeStretch (http://avisynth.org/mediawiki/TimeStretch)