EVOB De/Multiplexers [Archive] - Page 21

Rectal Prolapse

11th March 2007, 19:55

My sound hardware is an onboard Realtek ALC888DD. It is spec'ed to play 24 bit audio.

I can play the 24 bit FLAC in GraphEdit using the Native FLAC File Source filter without any problems. EDIT: Not muxed into an mka file.

I also can tell AC3Filter to output in 24 bits to DirectSound without any issues. It may be possible that the hardware downsamples to 16 bit before output to the speakers, but I cannot verify that at the moment.

I may give Wavpack a try - the ability to add a delay is nice!

chros

11th March 2007, 20:02

I don't understand why you guys do the intermediate files (raw, wav) if you want to encode to AC3 ... (Am I missing something?)

Why don't you use an avisynth script and behappy + aften ?

daveidmx

11th March 2007, 21:10

OK, I'm still stuck a few pages back on my DD+ decoding. I'm still working on the King Kong HD-DVD that came with my player, and the following holds for both the UNILOGO and FEATURE.

(Orbitlee source filter: demuxed DD+) -> (Sonic Cinemaster Audio Decoder 4.2) = 0x80040217, filters cannot agree on a connection.
(Async file: demuxed DD+) -> (Sonic audio) = 0x80040217
(Async file: demuxed DD+) -> (Sonic HD Demuxer) = so far so good; -> (Sonic audio) = 0x80040217
(Async file: EVO--rebuilt or original) -> (Sonic HD Dumuxer) = so far so good; -> (Sonic audio) = 0x80040217

Did I miss something?

I am able to hook up Orbitlee's filter to the Intervideo Audio Decoder, but the output is always a stream of zero-bytes exactly the same length as the source stream. (Possibly because my trial period has expired?)

However I can't for the life of me seem to hook anything up to the Sonic audio decoder. :confused:

Any thoughts?

D

madshi

11th March 2007, 22:35

FLAC, Wavpack, LPCM, MLP, TrueHD and DTS-HD Master Audio should all sound exactly the same since they are all lossless.
Yes, I know. I meant to ask how file size compares between these lossless formats. Sorry for being unclear.

madshi

11th March 2007, 22:38

I don't understand why you guys do the intermediate files (raw, wav) if you want to encode to AC3 ... (Am I missing something?)

Why don't you use an avisynth script and behappy + aften ?
Maybe the reason is that I've no experience with avisynth scripting at all... :o Another reason is that behappy originally didn't support 640 bitrate encoding with aften. But I've just yesterday found out that there's a modded build which supports it.

Anyway, can you decode E-AC3 via an avisynth script? How would that script look like? Thanks!

woah!

11th March 2007, 22:49

Deckard2019

12th March 2007, 00:18

Does anybody know anything about Blu-Ray subtitles ?
Is there a chance to extract them with xport ?

Chumbo

12th March 2007, 00:58

just build your sound graph and save it. then use this :

SetMemoryMax(64)
DirectShowSource("SOUND.GRF", video=false)

now just load that script into behappy and setup your aften options or just pick the preset one. i do the same thing except i go to aac instead.
Occasionally, the process may end in an exception. I found that if you add the fps parameter, the exception goes away.

For example: DirectShowSource("yourfile.grf", fps=x, video=false). Where x is your source fps, i.e., 23.976 or 29.970, etc.

behded

12th March 2007, 05:11

There are two reasons to force me use the intermediate raw and wav files:
1. the channel mapping. This maybe solved by some "matrix filter" which I don't know where to get. Can some one tell me please.
2. I always get the
"Error: System.IO.IOException: The pipe has been ended." error in the middle of the conversion in behappy, if I try to go from .grf directly to ac3. I don't know a way to solve this. I'm using the v0.06 of aften.I have the fps setting in the avs.

By the way, it seems the Sonic Audio Decoder 4.2 does not work under Vista. Can some one confirms this?

Occasionally, the process may end in an exception. I found that if you add the fps parameter, the exception goes away.

For example: DirectShowSource("yourfile.grf", fps=x, video=false). Where x is your source fps, i.e., 23.976 or 29.970, etc.

Chumbo

12th March 2007, 05:19

...2. I always get the
"Error: System.IO.IOException: The pipe has been ended." error in the middle of the conversion in behappy, if I try to go from .grf directly to ac3. I don't know a way to solve this. I'm using the v0.06 of aften.I have the fps setting in the avs.
...
You need the latest (build 449) of Aften. You should also grab the latest build of BeHappy. I believe both are available in the BeHappy thread. There's also a separate aften 0.06 thread. The latest build of BeHappy includes the latest aften.exe so that should take care of your problem.

Rectal Prolapse

12th March 2007, 06:21

daveidmx: Maybe you need to reregister both filters?

Also, check your audio configuration - make sure windows is set to 5.1 mode. Then, check the Cinemaster Decoder control panel and make sure it is also set to 5.1. Good luck!

I just gave WavPack a try - I can't get playback to work at all - FFDShow Audio Decoder completely hangs and everything stops - as if in a permanently paused state.

Those of you who got FLAC and WavPack working, what version of audio decoder are you using? What does the filter graph look like? I am stumped - maybe my FFDShow version is borked for lossless audio playback.

I'm using the latest (March something) Haali Media Splitter, with FFDShow-tryouts version 964. I used mkvtoolnix 2.0.2. I also used MediaCoder 0.5.1 to encode FLAC, WavPack, and FAAC. Only FAAC works flawless for me. FLAC almost works but with audio glitches (and no audio delay with mkvtoolnix).

I am sooooo close. :)

MichalHabart

12th March 2007, 07:19

Yes, of course, I'll do that. I'll also try to use Chumbo's suggestion to use the AC3Filter instead of ffdShow.

I'm always using 640 for 5.1 audio tracks. For 2.0 audio tracks I'm using 384, unless the E-AC3 track has a higher rate than that. In that case I'm using 640, too. I guess I can add a parameter, but is there anybody who would want to use 448, anyway? I mean we want best possible quality, don't we?

Thank you very much. I just wanted to know what bitrate do you use. Now when i know that it is 640kbps, i am quite happy with that :)

behded

12th March 2007, 08:34

I made sure I'm using the v449 of aften. My BeHappy is version 0.1.9.5241. Seems to be the latest version I can get from http://www.gotdotnet.com/workspaces/workspace.aspx?id=1bb59ddf-901b-43a5-bd54-b0999e8e223e
Is there any new download site, since the workspace is being phased out by M$.

I'm still getting the following error close to the end of the conversion:

Error: System.IO.IOException: The pipe has been ended.

at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
at System.IO.FileStream.Write(Byte[] array, Int32 offset, Int32 count)
at BeHappy.Encoder.encode()

You need the latest (build 449) of Aften. You should also grab the latest build of BeHappy. I believe both are available in the BeHappy thread. There's also a separate aften 0.06 thread. The latest build of BeHappy includes the latest aften.exe so that should take care of your problem.

orbitlee

12th March 2007, 11:34

Pelican9

12th March 2007, 11:49

For AC3 and DTS audio, DD+ source with sonic audio decoder 4.2 still can't give me reliable result, I give up, so simply DON'T use the source filter to handle AC3 and DTS. There are so many mature choices, right? :cool:

It could be better to rename this filter to DD+ source... :)

madshi

12th March 2007, 11:50

An updated version of DD+ source filter
http://www.sendspace.com/file/1vrahm

The first version of DD+ source filter only accepts bsid=16 DD+ audio, but yesterday I found a DD+ audio which has bsid!=16. DD+ source filter refuses to open this audio track.
Actually DD+ spec allows bsid=11~16. Should read spec more carefully :stupid:
Also, register the extension name .ec3

For AC3 and DTS audio, DD+ source with sonic audio decoder 4.2 still can't give me reliable result, I give up, so simply DON'T use the source filter to handle AC3 and DTS. There are so many mature choices, right? :cool:
Thank you!!

One little question to you, since you seem to have some experience with modding open source filters: Would it be much work for you to mod the AC3Filter so that it would not try to decode DD+? Right now when switching between DD+ and AC3 movies I always have to change preferences in MPC. If AC3Filter wouldn't try to decode DD+, that would allow a more comfortable setup.

orbitlee

12th March 2007, 12:36

It could be better to rename this filter to DD+ source... :)

After we find a reliable audio decoder, I'll be back :rolleyes:

Thank you!!

One little question to you, since you seem to have some experience with modding open source filters: Would it be much work for you to mod the AC3Filter so that it would not try to decode DD+? Right now when switching between DD+ and AC3 movies I always have to change preferences in MPC. If AC3Filter wouldn't try to decode DD+, that would allow a more comfortable setup.

Have you tried sonic audio decoder 4.2 to decoder AC3? (of course, not with the DD+ source filter ). I found haali splitter works with sonic audio decoder well (I think haali knows the directshow tricks :)

For ac3filter, currently I'm trying to build ac3filter, and check whether it could be configured(or patched) to decode LPCM 5.1 and 7.1 from Blu-ray properly. I will check how can I disable the not-working DD+ decoding. But since ac3filter is still being actively supported (http://ac3filter.net/forum/), I think it will be even better to request ac3filter developer to support DD+ and LPCM 5.1/7.1.

madshi

12th March 2007, 13:38

Have you tried sonic audio decoder 4.2 to decoder AC3?
I sometimes need passthrough over optical out and sometimes not. The AC3Filter lets me switch that on/off beautifully in the filter properties at runtime. With the Sonic decoder I can probably configure it from outside by using the external settings GUI. That's less comfortable and IIRC when I tried passthrough with the Sonic decoder it crashed for me.

For ac3filter, currently I'm trying to build ac3filter, and check whether it could be configured(or patched) to decode LPCM 5.1 and 7.1 from Blu-ray properly. I will check how can I disable the not-working DD+ decoding. But since ac3filter is still being actively supported (http://ac3filter.net/forum/), I think it will be even better to request ac3filter developer to support DD+ and LPCM 5.1/7.1.
Well, the last post of the admin was from October 2006, as far as I can see... :(

zgx

12th March 2007, 15:37

Those of you who got FLAC and WavPack working, what version of audio decoder are you using? What does the filter graph look like? I am stumped - maybe my FFDShow version is borked for lossless audio playback.

I'm using the latest (March something) Haali Media Splitter, with FFDShow-tryouts version 964. I used mkvtoolnix 2.0.2. I also used MediaCoder 0.5.1 to encode FLAC, WavPack, and FAAC. Only FAAC works flawless for me. FLAC almost works but with audio glitches (and no audio delay with mkvtoolnix).

I am sooooo close. :)To play a stand-alone FLAC you can use foobar2000. If you mux it into a MKV container you need a directshow filter. There are two filters available. You can read more about it here: http://www.inmatrix.com/zplayer/formats/flac.shtml.

I don't think you need to use ffdshow at all to decode FLAC. I use Zoomplayer and have setup "CoreFLAC Decoder v0.4" to decode FLAC and then I use AC3Filter after CoreFLAC so I can adjust levels and other things.

clsid

12th March 2007, 16:00

I just gave WavPack a try - I can't get playback to work at all - FFDShow Audio Decoder completely hangs and everything stops - as if in a permanently paused state.
Wavpack support in ffdshow is incomplete. Use CoreWavpack instead.
http://coreforge.org/frs/?group_id=28

Rectal Prolapse

12th March 2007, 16:06

Thanks for pointing me in the right direction, zgx EDIT: and clsid. I couldn't get the original FLAC decoder to connect after Haali's splitter (different media subtype), but your alternatives should help me out!

I'll see if there is a separate Haali-compatible decoder for WavPack too. I really could use the delay after demuxing HD-DVDs -somehow this delay information is lost after the conversion!

I suppose I can find a WAV utility to add the required silence, but again that is outside my expertise. :(

BTW, I've been experimenting with WMA Lossless. The file size is much larger than expected, and it doesn't help that the conversion tool (using WMP10 I think) has a 4 GB limit with WAVs - it will only convert up to 74 minutes and 13 seconds, and not tell me it ignored the rest! Blah.

Oh well - I can't mux a WMA Lossless stream into an MKV anyways. Although MPC can load it as an external soundtrack and works VERY nicely I might add - until you hit the silence after 74:13.

Also, it appears that gdsmux does not support FLAC. When I try to mux it in, it thrashes around for a few seconds on the hard disk, and then it does nothing forever after that - and leaves me with a 0 byte MKV file. So I guess you must use MKVToolNix for the time being.

Rectal Prolapse

13th March 2007, 00:07

Darth Pinous said:

You should try WavPack instead of FLAC. You can delay a WavPack soundtrack in mkvmerge.

Hmmm, this didn't work for mkvmerge 2.0.2. The delay setting was ignored in playback when using WavPack, which I just managed to get working in an mkv, so far *fingers crossed*.

I will report back after trying gdsmux.

Rectal Prolapse

13th March 2007, 00:15

http://forum.doom9.org/showthread.php?p=958726#post958726

mkvmerge should add silence, but it doesn't support this for all audio formats (only for AC3, MP3, Vorbis if I'm not mistaken). At the moment it creates weird sound for AC3 because it does the wrong thing.

Well, there you go. :(

EDIT: It appears to work with gdsmux though....

Darth Pinous

13th March 2007, 11:06

Darth Pinous said:

Hmmm, this didn't work for mkvmerge 2.0.2. The delay setting was ignored in playback when using WavPack, which I just managed to get working in an mkv, so far *fingers crossed*.

I will report back after trying gdsmux.

I just noticed that... I will try to add delay in BeHappy, when the Wavpack soundtrack is created...

madshi

13th March 2007, 11:11

Did anyone compare how a 640kbit E-AC3 -> AC3 converted audio track sounds compared to a 448kbit DVD original audio track? Do they sound roughly identical or is one better than the other?

Thanks!

MichalHabart

13th March 2007, 11:36

Did anyone compare how a 640kbit E-AC3 -> AC3 converted audio track sounds compared to a 448kbit DVD original audio track? Do they sound roughly identical or is one better than the other?

Thanks!

I did not do any deep comparision but eac3 should always sound better then dvd ac3 especially because it is from better source and has better dynamic then ordinary ac3

madshi

13th March 2007, 11:45

I did not do any deep comparision but eac3 should always sound better then dvd ac3 especially because it is from better source and has better dynamic then ordinary ac3
Sure. But my question was about reencoded E-AC3. You know, every lossy reencoding hurts quality. So it's not clear to me which sounds better:

(1) E-AC3 to 640kbit AC3
+ advantage: final AC3 is 640kbit instead of 448kbit
- disadvantage: encoded + reencoded, both lossy

(2) 448kbit AC3 from DVD
+ advantage: encoded only once
- disadvantage: final AC3 is 448kbit instead of 640kbit

So which sounds better? I'm not very experienced in comparing soundtracks. So I'm not sure whether I can trust my ears. I'd really love to hear some comments about what you guys think.

MichalHabart

13th March 2007, 12:47

Sure. But my question was about reencoded E-AC3. You know, every lossy reencoding hurts quality. So it's not clear to me which sounds better:

(1) E-AC3 to 640kbit AC3
+ advantage: final AC3 is 640kbit instead of 448kbit
- disadvantage: encoded + reencoded, both lossy

(2) 448kbit AC3 from DVD
+ advantage: encoded only once
- disadvantage: final AC3 is 448kbit instead of 640kbit

So which sounds better? I'm not very experienced in comparing soundtracks. So I'm not sure whether I can trust my ears. I'd really love to hear some comments about what you guys think.

Definitely ac3 transcoded from eac3

zgx

13th March 2007, 13:02

Sure. But my question was about reencoded E-AC3. You know, every lossy reencoding hurts quality.You are right - every lossy encode hurts quality. It's not obvious that the 640 Kbps AC3 reencode will sound better then the 448 Kbps AC3 from a DVD.

Lossless Source -> 640 Kbps E-AC3 -> 640 Kbps AC3
Lossless Source -> 448 Kbps AC3

I'm not sure if I could hear any difference in a ABX test but I don't want to take any chances so I prefere to play the E-AC3 track directly. There is no need to reencode it. Configure your favorite player (ZoomPlayer or MPC for example) to decode "AC3" with "Sonic Cinemaster Audio Decoder 4.2". Then put AC3Filter "behind" the Sonic filter and you should be able to decode E-AC3 tracks. Of course you can't use S/PDIF with E-AC3 unless you are fine with AC3Filter doing a realtime "reencode" to AC3.

You can also mux a E-AC3 tracks with your video in a MKV container.

madshi

13th March 2007, 13:14

Definitely ac3 transcoded from eac3
Are you guessing or did you do a real comparison?

You are right - every lossy encode hurts quality. It's not obvious that the 640 Kbps AC3 reencode will sound better then the 448 Kbps AC3 from a DVD.

Lossless Source -> 640 Kbps E-AC3 -> 640 Kbps AC3
Lossless Source -> 448 Kbps AC3

I'm not sure if I could hear any difference in a ABX test but I don't want to take any chances so I prefere to play the E-AC3 track directly. There is no need to reencode it. Configure your favorite player (ZoomPlayer or MPC for example) to decode "AC3" with "Sonic Cinemaster Audio Decoder 4.2". Then put AC3Filter "behind" the Sonic filter and you should be able to decode E-AC3 tracks. Of course you can't use S/PDIF with E-AC3 unless you are fine with AC3Filter doing a realtime "reencode" to AC3.

You can also mux a E-AC3 tracks with your video in a MKV container.
Well, my receiver doesn't have HDMI inputs and I don't have a 5.1 analog connection between PC and receiver (too far away), so SPDIF is the only choice I have. That's why I'm asking.

I do plan to mux both the E-AC3 (for future use) and an AC3 (for compatability) track into my MKVs. The big question is whether I should mux a DVD ripped AC3 track or whether I should mux an E-AC3 -> AC3 converted track. That's really a tough decision!

JeffAlso

13th March 2007, 14:37

Lossless Source -> 640 Kbps E-AC3 -> 640 Kbps AC3
Lossless Source -> 448 Kbps AC3
I'm no expert here, but I would highly suspect that the amount of quality loss from a reencode would be significantly less than the quality loss suffered from lowering the bitrate by 30%.

MichalHabart

13th March 2007, 15:36

Are you guessing or did you do a real comparison?

It is based on listening comparision done by JnZ

madshi

13th March 2007, 16:02

It is based on listening comparision done by JnZ
Do you have a link to that comparison? I read all JnZ posts in this thread and didn't find a comparison. Thanks!

Ashraf-Khan

13th March 2007, 16:03

hello all,

nice forum & helpful!!

so, let me say that i love win media center.
for that reason i want to copy my 13+ hd-dvds (just the pure movie evos) on a new 500 gig :cool: hdd.
yesterday i just installed sonic decoder pack and it worked!

but only on one pc.

on the other it won't work whether just playing in mce or in graphedit
(audio dec. v. 4.2.xxx.102 , video dec. xxx140 & demuxer xxx61)

furthermore i can't connect for example demuxer with audio dec! (no "zwischenfilter")

could you please post your ver. numbers and if the filters can be connected so we can figure out which versions work & which is the newest!

olli

zgx

13th March 2007, 16:06

I do plan to mux both the E-AC3 (for future use) and an AC3 (for compatability) track into my MKVs. The big question is whether I should mux a DVD ripped AC3 track or whether I should mux an E-AC3 -> AC3 converted track. That's really a tough decision!Mux the E-AC3 track and let AC3Filter do realtime re-encoding to AC3 to output over S/PDIF. Might not sound as good as a professional AC3 re-encode but then you will have the E-AC3 intact for future use.

I'm no expert here, but I would highly suspect that the amount of quality loss from a reencode would be significantly less than the quality loss suffered from lowering the bitrate by 30%.You are probably right.

MichalHabart

13th March 2007, 16:15

Do you have a link to that comparison? I read all JnZ posts in this thread and didn't find a comparison. Thanks!

No, i have our conversation only in history of QIP :)

Rectal Prolapse

13th March 2007, 16:30

DarthPinous - BeHappy can add delays? I didn't know that - and it can encode in WavPack? Well, that would save me a lot of steps! (and hard drive space).

madshi

13th March 2007, 16:38

Mux the E-AC3 track and let AC3Filter do realtime re-encoding to AC3 to output over S/PDIF. Might not sound as good as a professional AC3 re-encode but then you will have the E-AC3 intact for future use.
This way I'd have a MKV which would be perfect for future use. But it would be suboptimal for today's use. I'm not satisfied with that. I want to have the best possible AC3 track in my MKV, right next to the original E-AC3 track.

No, i have our conversation only in history of QIP
Ok, thanks!

JnZ

13th March 2007, 17:15

No, i have our conversation only in history of QIP :)
Well guys, I write it something like this:
I compared only output formats: AC3 vs DTS vs OGG, all in blind ABX test on my headphones Koss Porta Pro...soundcard M-Audio Revo 5.1 (muxed 5.1 to 2.0 first).

I test this:
- AC3 from BeSweet 448,640kbps
- DTS from SurCode 1536,768kbps
- OGG from AoTuV Q6,Q5,Q4,Q3,Q2,Q1.

Definitely, I must say I can't hear any diferences between this formats. Only in case DTS 768 I hear very-very-small difference (I suspects SurCode for producing bad crap). I've been only surprised, OGG produced very good quality in small, acceptable biterate...I can't hear any difference even at biterate around 180-200kbps!!! I hear margin, until I lowered biterate to aprox 160kbps.

Other thing: When I do ABX tests, I must gain volume in AC3 about 6-8dB compared DTS and OGG formats. OGG and DTS had almost the same level of volume...this is very good, because OGG sounds like DTS, not like crappy silent AC3. So I choose for me OGG. But pros for AC3: when boosted, it sounds like DTS or OGG (I can't hear any difference).

I can't make hear tests DD+ vs AC3, coz in my region, I have PAL DVD's which have different pitch of audio track...and transcoding via besweet 25->23.976fps is another quality lost. But if I can say in my fleeting test...AC3 from DD+ sounds better. ;)

J.

madshi

13th March 2007, 18:12

Thanks, JnZ.

Darth Pinous

13th March 2007, 18:38

DarthPinous - BeHappy can add delays? I didn't know that - and it can encode in WavPack? Well, that would save me a lot of steps! (and hard drive space).

There's an option, but I don't know if it works. I will try a Wavpack reencode with delay tonight, and make a reply then.

Friedl2000

13th March 2007, 19:10

Hi,

how to display the desired subtitle after a rebuild.

Wanna do 2 Tracks in the new evo + 2 subtiles.

How can i switch the audio and subs during playback and in which program ? PowerDVD7.2 ????

many thx

Rectal Prolapse

13th March 2007, 21:37

Darth, I've played around with BeHappy - it appears that adding delay works - makes sense, given that it uses AVISynth!

Some annoyances: BeHappy (and MediaCoder!) downsamples everything to 16 bit. I was wondering why my WavPack and FLAC encodes were so small! 7.9 gig 5.1 multichannel 24 bit PCM soundtrack gets compressed down to 1.3 gig FLAC or WavPack in 16 bits!

When I run WavPack directly from the command line, feeding it the 7.9 gig WAV soundtrack through stdin, the file size is more than double the 16 bit one - 3.7 gigs. Ouch. :)

I don't know why MediaCoder and BeHappy downsample to 16 bit, when the encoders can clearly take 24 bit samples. Maybe it is a limitation of AVISynth?

Rectal Prolapse

13th March 2007, 22:09

Darth Pinous

13th March 2007, 22:18

zgx

13th March 2007, 22:28

It appears that FLAC command-line has an output (and input?) filesize limit of 2 GB - making it inappropriate for lossless 24 bit multichannel tracks!

I'm not sure if there are alternative FLAC encoders that can handle 24 bits, 48000 KHz multichannel audio.In http://flac.sourceforge.net/changelog.html#flac_1_1_3 you can read "Large file (>2GB) support everywhere".

But I also know that flac.exe can't handle huge wave files. The problem I think is with the WAV files. "The WAV format is limited to files that are less than 4 GiB in size, due to its use of a 32 bit unsigned integer to record the file size header (some programs limit the file size to 2 GiB)."

But if you use MediaCoder to encode your FLAC files you can use as large WAV files as you like so 24 bit multichannel tracks should be no problem.

idamien

14th March 2007, 01:11

Does anyone know of any free/open source DD+ decoders already around? Will AC3Filter support the new format?

Applecore

14th March 2007, 01:21

hey guys

ive been following your thread for a while now and have been experimenting with my hd backups. you keep mentioning this sonic audio decoder 4.2. i cant find that anywhere? theres no mention of it on any site besides this one :)
if you could help me out id greatly appreciate it.

idamien

14th March 2007, 02:00

Hello Applecore and welcome!

The Sonic Audio Decoder is part of Sonic Scenarist 4, I think. Sonic Scenarist is a professional SD/HD-DVD authoring software that costs a few hundred dollars (guessing)...

Rectal Prolapse

14th March 2007, 02:09

zgx, unfortunately mediacoder seems to downconvert the PCM tracks to 16 bit. Or more specifically, it appears that mplayer is doing it.

Another problem - suddenly my channels are in the wrong order after encoding to FLAC/WAVPACK! I don't get it - it was working fine yesterday. I wish I could have more control over MediaCoder's configuration so that I can re-arrange the channels.

I know that the WAV file I generated using Sox is correct - it works great with BeHappy.

I am stumped - why did it change suddenly? How come it didn't happen before? I'm reading the massive mplayer documentation (what a mess!) and it appears to have very funny ideas on channel order - when I play the WAV with it they are all in the wrong positions compared to all the other players out there.

Anyways, back to more hacking - these GUI front end tools are a mess - I would rather hack around with AVISynth instead. :)