View Full Version : FFDShow isn't decoding 16bit 44.1kHz DTS tracks for me
Ditto666
3rd October 2009, 21:55
I set the default for DTS decoding to FFDShow. I even added it in the "External Filters" of Media Player Classic and prioritized it there, also selecting it for DTS media. For some reason, it only defaults to decoding 24bit 48kHz streams. Otherwise, it defaults to AC3Filter. I went ahead to disable AC3Filter but if I do that, it defaults to nothing and then all I get is static. I've looked through every option that i thought was relevant in FFDShow but maybe I missed something. Anybody have any idea as to why this is happening? Thanks :D
UPDATE: It has to do with 44.1kHz frequencies. 16bit 48kHz DTS streams worked.
Ditto666
6th October 2009, 04:24
I really can't figure it out... Can somebody at least tell me what to maybe try if they don't exactly know what the problem is? Really, nobody knows?
-This place does kinda seem motionless though, so maybe that's why... I thought Doom9 was a very active forum, heh XD...
*Maybe this should be posted under a different section?..
Inspector.Gadget
6th October 2009, 05:31
Upload a sample. Let's see if we can reproduce the problem.
Midzuki
6th October 2009, 05:40
Also, tell us what is the version of ffdshow you're using.
BTW, if AC3Filter works correctly, why not use it? :)
Ditto666
6th October 2009, 15:12
Also, tell us what is the version of ffdshow you're using.
BTW, if AC3Filter works correctly, why not use it? :)
I have FFDShow revision 2527. I've always been using AC3Filter. I'm trying to use FFDShow now because I can upmix to 7.1 surround with it.
Ditto666
6th October 2009, 15:15
Upload a sample. Let's see if we can reproduce the problem.
Well, I'm pretty sure that it won't be like that for most other people. It's basically with any file I've tried though. Do you want a sample because you don't have those type of files? If that's the case, then sure, I'll upload the samples :cool:
Inspector.Gadget
6th October 2009, 16:52
Well, there's either a problem with your config or with the files you're playing. This will help us figure out which.
Also, you don't need both ffdshow and AC3filter. You're safe with one or the other, as far as audio goes.
Ditto666
7th October 2009, 00:32
Well, there's either a problem with your config or with the files you're playing. This will help us figure out which.
Also, you don't need both ffdshow and AC3filter. You're safe with one or the other, as far as audio goes.
I've tried like every file... It's not the files. And the files work with AC3Filter. It's very obviously not them.
I like AC3Filter and I need both for multiple reason. One has to upmix stereo audio files and the other has to upmix 5.1 to 7.1. It's because of what I explained in another thread. I will quote it. And also, if I uninstall it now, I will just get static, as I said earlier in this thread.
"I actually just finished trying FFDShow Audio and I set up the 7.1 configuration. I like it BUT, there are a few problems and maybe a bias that I need to clear up...
There is no way to save the Matrix Mixer preset that I created. And even if there was, it'd still be annoying to have to go into the settings and choose between the stereo and 5.1ch presets that I'd have. I know that I CAN save all the settings overall but going about it that way wouldn't be efficient or too effortless either. As it stands now anyway, I'd have to manually change the configuration each time I decided to switch between different listening sources, stereo and 5.1...
Also, I think I might've just had a bias towards FFDShow Audio because I initially (long ago) couldn't find how to output audio in 5.1 but was able to in AC3Filter instantly after I installed it. I grew to like AC3Filter in every aspect. One of those aspects was always just the way it upmixed stereo audio. Now that I dealt with the Matrix Mixer though, I realized that this upmixture is nothing special and that nothing past the use of the mixer is happening. The default settings in FFDShow and AC3Filter are exactly the same in the mixer. So finally, that being said, is there actually any difference in the decoding of MP3, AAC, AC3, and DTS audio (in terms of anything - quality, stereo upmixing, sound redirection maybe, etc...)? If so, what are the actual technical differences, which decoder do you prefer, and why? Just uh, please don't answer something along the lines of, "FFDShow PWNS! AC3Filter FTL!".. yeah...
Didn't realize I'd have this much to ask.. Big box of virtual cookies to anybody that takes the time to answer this But seriously, thanks very much."
*Also, update in main post.
Midzuki
7th October 2009, 03:43
First of all, I'd like to say that I too use both AC3Filter and ffdshow audio decoder+processor. :)
Before recommending an update to your ffdshow ( mine is build 3008, and it's outdated already :D ), : have you checked whether those 44.1kHz DTS files are "SPDIF-wrapped"? Usually, DTS @ 44.1kHz is a .WAV file containing zero-padded DTS "noise", and it's used for multichannel Audio CDs. AC3Filter can detect and decode AC3/DTS/MP2 wrapped in a SPDIF stream, whereas ffdshow, SFAIK, cannot.
You can convert ".wavved DTS" to pure DTS with bsconvert.exe
(part of the AC3Filter Tools package).
Ditto666
7th October 2009, 06:43
First of all, I'd like to say that I too use both AC3Filter and ffdshow audio decoder+processor. :)
Before recommending an update to your ffdshow ( mine is build 3008, and it's outdated already :D ), : have you checked whether those 44.1kHz DTS files are "SPDIF-wrapped"? Usually, DTS @ 44.1kHz is a .WAV file containing zero-padded DTS "noise", and it's used for multichannel Audio CDs. AC3Filter can detect and decode AC3/DTS/MP2 wrapped in a SPDIF stream, whereas ffdshow, SFAIK, cannot.
You can convert ".wavved DTS" to pure DTS with bsconvert.exe
(part of the AC3Filter Tools package).
Well, yes, that is what they are. I thought all that had to be done was rename them to ".dts" instead of ".wav". Apparently that's not the case. Thanks for the recommendation, but I don't think I want to convert all my hundreds of files, even if there isn't quality loss. Thank you very much though, seriously; you actually gave me the answer I thought I won't find. Guess no way to get this to work directly then.. meh...
Maybe AC3Filter will have an update soon for 8 channel support in the Matrix Mixer at least. I was also thinking to update FFDShow but from what you said, looks like that wouldn't change anything. Might you know of just some individual Matrix Mixer that can be used to further process the information decoded by AC3Filter?
tebasuna51
7th October 2009, 11:42
FFDShow isn't decoding 16bit 44.1kHz DTS tracks for me
I think you must chage the title of your post because you don't have DTS tracks but fake dtswav files.
These files have a wrong wav header to deceive burn CD software and, after burn the CD, must be deleted, or converted to true dts files if you want compatibility with all players and save storage space, the true DTS file is 7/8 in size than the fake dtswav file equivalent.
clsid
7th October 2009, 11:55
If it is DTS inside WAV, then you could try to also enable "Uncompressed" audio in ffdshow. That is a workaround that usually makes them work with ffdshow.
Midzuki
7th October 2009, 14:01
Might you know of just some individual Matrix Mixer
that can be used to further process the information decoded by AC3Filter?
Sadly I am not aware of any other "matrix mixers" than the ones from
AC3Filter and ffdshow, respectively. However, it's indeed possible to decode
a 5.1 audio source with AC3Filter and upmix the uncompressed output
with ffdshow's Audio Processor. Here is the method that's worked for me:
1) set AC3Filter to decode PCM, AC3, DTS, and DVD ;
2) check the radio-button "Prefer AC3Filter" ;
3) use RadLight Filter Manager and set the merit of ffdshow
"Audio Decoder" to 00600002, and the merit of ffdshow "Audio Processor" to
00800001; besides, set the "Codecs" section of the Audio Decoder to handle
uncompressed audio ;
4) Untick all checkboxes in the left panel of ffdshow's Audio Decoder
configuration applet ;
5) Check the item "Mixer" in the config window of ffdshow Audio Processor,
and tweak as needed ; also, make sure the "Output" settings say
«Connect to any filter» ;
6) TEST! these DirectShow settings in Media Player Classic or
in Windows Media Player ;
HTH.
Ditto666
7th October 2009, 21:54
I think you must chage the title of your post because you don't have DTS tracks but fake dtswav files.
These files have a wrong wav header to deceive burn CD software and, after burn the CD, must be deleted, or converted to true dts files if you want compatibility with all players and save storage space, the true DTS file is 7/8 in size than the fake dtswav file equivalent.
Oh wow. I never knew that. I wouldn't necessarily call them fake though because they produce the same quality.
If it is DTS inside WAV, then you could try to also enable "Uncompressed" audio in ffdshow. That is a workaround that usually makes them work with ffdshow.
Wow.. That worked. I can't believe it. Thank you.
Ditto666
7th October 2009, 22:01
Sadly I am not aware of any other "matrix mixers" than the ones from
AC3Filter and ffdshow, respectively. However, it's indeed possible to decode
a 5.1 audio source with AC3Filter and upmix the uncompressed output
with ffdshow's Audio Processor. Here is the method that's worked for me:
1) set AC3Filter to decode PCM, AC3, DTS, and DVD ;
2) check the radio-button "Prefer AC3Filter" ;
3) use RadLight Filter Manager and set the merit of ffdshow
"Audio Decoder" to 00600002, and the merit of ffdshow "Audio Processor" to
00800001; besides, set the "Codecs" section of the Audio Decoder to handle
uncompressed audio ;
4) Untick all checkboxes in the left panel of ffdshow's Audio Decoder
configuration applet ;
5) Check the item "Mixer" in the config window of ffdshow Audio Processor,
and tweak as needed ; also, make sure the "Output" settings say
«Connect to any filter» ;
6) TEST! these DirectShow settings in Media Player Classic or
in Windows Media Player ;
HTH.
Yeah, that seems like it would work - very cool. I have most of those settings like that already besides the merits. One question though - one that I've asked before and think you might know some things about.
I've always had this bias, all for AC3Filter. But really, FFDShow seems to have all the things AC3Filter has and more. Do they work any differently when speaking of audio quality/clarity? If they're almost identical, why not just use FFDShow? Thanks.
Midzuki
8th October 2009, 01:00
@ Ditto666: actually, I can only say, AC3Filter is a "hard habit to break". :o I'm still using version 1.46. :D And when I knew the most recent versions were full of new bugs, so to speak, ... Oh well.
Regarding what I wrote one post ago: AC3Filter does not have a resampling module, so I use ffdshow's one. :cool: OTOH, in ffdshow, the MLP decoder depends on the AC3 decoder. :eek: To my brains, that's unwise, period. The old Netscape Communicator had a similar design flaw --- the CSS interpreter didn't work at all if the JavaScript engine was disabled, and if the web designer had chosen to store a big .js in a .jar, the browser was forced to start its Java engine only in order to decompress one archive. :mad:
Ditto666
8th October 2009, 02:37
@ Ditto666: actually, I can only say, AC3Filter is a "hard habit to break". :o I'm still using version 1.46. :D And when I knew the most recent versions were full of new bugs, so to speak, ... Oh well.
Regarding what I wrote one post ago: AC3Filter does not have a resampling module, so I use ffdshow's one. :cool: OTOH, in ffdshow, the MLP decoder depends on the AC3 decoder. :eek: To my brains, that's unwise, period. The old Netscape Communicator had a similar design flaw --- the CSS interpreter didn't work at all if the JavaScript engine was disabled, and if the web designer had chosen to store a big .js in a .jar, the browser was forced to start its Java engine only in order to decompress one archive. :mad:
I think I'm going to go insane. I don't know why I'm doing this to myself. I've been at this for 3 hours straight. I've been trying to calibrate my speakers to perfection even though I earlier thought they were. I was trying different things in the Matrix Mixer to achieve 7.1 surround. And something just wasn't right. I heard so much that I wasn't even sure any more after I did the following. I turned back to AC3Filter, with only the 5.1 surround and the sound quality sounded much less suppressed. Maybe it's a combination of the receiver or maybe not. I don't know. I really hope it's not true cause then everything would've been to waste. My OCD is killing me and I can't enjoy any of my music. That's it, I'm making a damn poll...
Ditto666
9th October 2009, 06:15
Sadly I am not aware of any other "matrix mixers" than the ones from
AC3Filter and ffdshow, respectively. However, it's indeed possible to decode
a 5.1 audio source with AC3Filter and upmix the uncompressed output
with ffdshow's Audio Processor. Here is the method that's worked for me:
1) set AC3Filter to decode PCM, AC3, DTS, and DVD ;
2) check the radio-button "Prefer AC3Filter" ;
3) use RadLight Filter Manager and set the merit of ffdshow
"Audio Decoder" to 00600002, and the merit of ffdshow "Audio Processor" to
00800001; besides, set the "Codecs" section of the Audio Decoder to handle
uncompressed audio ;
4) Untick all checkboxes in the left panel of ffdshow's Audio Decoder
configuration applet ;
5) Check the item "Mixer" in the config window of ffdshow Audio Processor,
and tweak as needed ; also, make sure the "Output" settings say
«Connect to any filter» ;
6) TEST! these DirectShow settings in Media Player Classic or
in Windows Media Player ;
HTH.
I've established that AC3Filter does in fact create better quality than FFDShow. Will this method that you mentioned maintain that quality and just utilize FFDShow's Matrix Mixer?
Also, with the settings I have in the attached picture (or here: http://img16.imageshack.us/img16/8865/ffdshowaudiomatrixmixer.jpg), as an example - is putting a number in the intersection of Left and Side Left actually taking that amount away from left and putting it into Side Left or is it replicating the sound? In this 5.1->7.1 upmix, I'm trying to preserve the 5.1 and just add to the sides by replicating some of the sound from the front and back. Is that what's being done here or not?
And finally, regarding RadLight Filter Manager, when I try to change the merit of FFDShow Audio, it gives me an error saying "Cannot update merit".
tebasuna51
9th October 2009, 13:25
I think you are wrong.
- Ac3Filter decoder isn't better than ffdshow
- Your matrix 5.1 -> 7.1 isn't correct.
C' = 0.05xFL + 1.1xC + 0.05xFR
SL' = 0.3xFL - 0.3xFR + 0.6xBL - 0.6xBR
SR' = -0.3xFL + 0.3xFR - 0.6xBL + 0.6xBR
LFE' = 0.1xFL + 0.1xFR + LFE
I can't understand for what you change C and LFE channels, and the mixture in Side channels can be anything curious.
Must be:
C' = C
SL' = 0.7xBL
SR' = 0.7xBR
BL' = 0.7xBL
BR' = 0.7xBR
LFE' = LFE
Ditto666
9th October 2009, 22:03
I think you are wrong.
- Ac3Filter decoder isn't better than ffdshow
- Your matrix 5.1 -> 7.1 isn't correct.
C' = 0.05xFL + 1.1xC + 0.05xFR
SL' = 0.3xFL - 0.3xFR + 0.6xBL - 0.6xBR
SR' = -0.3xFL + 0.3xFR - 0.6xBL + 0.6xBR
LFE' = 0.1xFL + 0.1xFR + LFE
I can't understand for what you change C and LFE channels, and the mixture in Side channels can be anything curious.
Must be:
C' = C
SL' = 0.7xBL
SR' = 0.7xBR
BL' = 0.7xBL
BR' = 0.7xBR
LFE' = LFE
I didn't compare with these settings. I compared with the default, just regular 5.1 surround settings, where each channel just has 1:1.
Thing about me being wrong is, I wanted FFDShow to be better because that would make things a lot simpler for me. I was upset to hear when I played the same thing with AC3Filter that it sounded better. So it's not like my bias made me hear something that wasn't there. (EDIT: Just did a test with a friend, blind to all of this. He repeatedly confirmed AC3Filter as better on every song. I never told him which one is which.)
Anyway, regarding what I did, here's what I wanted to do and then you'll tell me how stupid what I did was and what I actually need to do:
1) I want to transfer a little bit from the front channels into the center (hence the .05's directed to the center)
2) I want to make the center channel a bit louder (hence the 1.1)
3) I want to mix 2/3 of the back channels with 1/3 from the front channels into the sides and somehow keep it leveled in terms of loudness (hence the numbers directed towards the side channels)
4) This may be stupid but I wanted to mix a bit from the front channels into the LFE. I basically just want more stuff to go into the bass, but, no, a higher bass redirection frequency is not what I want.
***I want #3 that way, under the assumption that the Matrix Mixer works the way I think it might. I'm not too sure how this works. I basically just want the side speakers to be like the crossover channels. So like, when a sound transfers from a back to a front speaker, it would go through the center first. Putting 0.6 in the intersection of Back Left and Side Left is obviously doing something else than just replicating 60% of that sound into the side... I never quite understand how it determines which sound is sent to the back (like in the 2ch->5.1ch upmixture).
clsid
9th October 2009, 22:25
You can't just add the fractions like that because the volume levels are measured in decibels. If I remember correctly you need to take the square root.
50% -> sqrt(0.500) = 0.707
66% -> sqrt(0.667) = 0.816
Ditto666
10th October 2009, 10:00
You can't just add the fractions like that because the volume levels are measured in decibels. If I remember correctly you need to take the square root.
50% -> sqrt(0.500) = 0.707
66% -> sqrt(0.667) = 0.816
Oh, I see... I don't understand how any of this works and why it's like that but ok, that's what I'll do >.< Hopefully my questions in the previous post will be answered :(
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