AAC 5.1 to AC3 SPDIF passthru with ffdshow problem

oddball · 21st February 2005, 20:45

AAC 5.1 to AC3 SPDIF passthru does not work. Audio stutters continuously. There appears to be a problem with libfaad2. Using Tremor on a Vorbis 5.1 file no problemo. Plays back as AC3 5.1 via SPDIF. But not AAC 5.1

Any ideas?

bond · 21st February 2005, 21:01

Quote:

Originally posted by oddball
AAC 5.1 to AC3 SPDIF passthru does not work. Audio stutters continuously. There appears to be a problem with libfaad2. Using Tremor on a Vorbis 5.1 file no problemo. Plays back as AC3 5.1 via SPDIF. But not AAC 5.1 Any ideas?

i assume you use he-aac? try lc-aac and report back

splitted and moved to audio forum, i think this deserves an own thread and not being lost in the ffdshow development thread

SeeMoreDigital · 21st February 2005, 21:33

Quote:

Originally posted by oddball
AAC 5.1 to AC3 SPDIF passthru does not work. Audio stutters continuously. There appears to be a problem with libfaad2. Using Tremor on a Vorbis 5.1 file no problemo. Plays back as AC3 5.1 via SPDIF. But not AAC 5.1

Which version of FFdshow did you install? I used Celtic Druid's 22Jan05 version.

It works perfectly with my 6Ch AAC-HE Terminator3 Mpeg4/AVC test file.

Cheers

Ana · 21st February 2005, 23:22

I had spdif passtrough problems with aac files too. My solution was resampling from 44.1khz to 48khz because my soundcard couldn't handle 44.1 khz via spdif properly. If I remember correctly nero makes only 44.1 khz aac-files...

oddball · 21st February 2005, 23:25

That worked! You rule!

Ana · 23rd February 2005, 11:40

Thanks.

BTW there is a way to do 48kHz aac files with foobar using nero encoder. http://forum.doom9.org/showthread.ph...hreadid=67746&

oddball · 24th February 2005, 06:12

If I was encoding them I would

BTW AC3filter RC5 has no problem (Maybe it is resampling without showing it is).

SunShock · 3rd March 2005, 09:39

Quote:

Originally posted by oddball

BTW AC3filter RC5 has no problem (Maybe it is resampling without showing it is).

Not true. My 44.1 KHz audio had issues with AC3Filter 1.01 RC5, while my 48KHz does not. Similarly, the resample option in ffdshow is what makes it work for me. I believe other people mentioned this sound card-specific issue in the AC3Filter RC thread.

oddball · 3rd March 2005, 23:08

I use an M-Audio Revolution 7.1 soundcard.

SeeMoreDigital · 3rd March 2005, 23:33

As I posted in another section of the forum, what I would like, is for FFdshow to offer a totally separate "AC3 Transcoding" application.

So if there's anybody here who can whip up a very basic/no thrills AC3 transcoding application, that's very light on processing power... please do so!

I thought maybe it could look/offer something like this: -

Cheers

oddball · 6th March 2005, 23:22

AT present I have enabled ffdshow for AC3, DTS and MP3 with AC3 and DTS being passed to SPDIF and MP3 passing untouched as analog. I use AC3Filter for AAC and Ogg to 5.1 and 2.0 SPDIF. It kinda works. But ffdshow audio allowing to route each seperately would be ideal Especially allowing 2.0 or 1.0 MP3 Ogg and AAC to pass as analog whilst 5.1 AAC and Ogg get encoded to AC3 and sent to SPDIF. I still have the 44Khz AAC issue in ffdshow and have to use resampling if I want to use it. But at present I have just disabled it and let AC3Filter handle AAC instead as it handles 44Khz AAC with no problems whatsoever (For me).

madman1980 · 1st July 2005, 23:38

I'm trying to play a file now with ac3 5.1 48khz (dd) and my receiver only says it gets a dpl signal (stereo).

How do I make ffdshow just use passthru? I don't want the signal to be messed with, just passed on to my receiver. I couldn't find any settings in ffdshow audio options that looked like it had anything to do with passthrough.

Using ffdshow 19062005, letting ffdshow do the audio (ac3filter shouldnt be necessary anymore, right?).

oddball · 3rd July 2005, 17:13

In the latest version there should be a drop down option next to the AC3 codec to select SPDIF.

madman1980 · 3rd July 2005, 21:18

That worked. Thanks

Rockaria · 5th July 2005, 23:30

Quote:

Originally Posted by SunShock

Not true. My 44.1 KHz audio had issues with AC3Filter 1.01 RC5, while my 48KHz does not. Similarly, the resample option in ffdshow is what makes it work for me. I believe other people mentioned this sound card-specific issue in the AC3Filter RC thread.

Yeah, the 48K only worked fine with ac3filter for my soundstorm/realtek with spdif passthrough or ac3 s/w dynamic encoding-passthrough.

As for the ffdshow audio, I couldn't enable both resampling(to 48K) and ac3 passthrough at the same time.
But when used with ac3filter together in the graphedit(ffdshow as resampling and ac3filter as ac3 encoding), it worked for me for 44.1K 5.1ch.

Maybe a playback syncing problem when ffdshow used alone?

E-Male · 6th July 2005, 05:18

i have used the ffdshow(resampling) and ac3filter(encoding) combination successfully in mediaplayerclassic using the ffilter-overides options

btw you can encode 48khz aac with besweet, just use commandline parameters with bsn (and NOT the config window)

Rockaria · 6th July 2005, 09:45

Glad to know that the mediaplayer classic has the advanced filter option.
I also was able to get 48K AACs in a step using foobar2k with a resampling plugin enabled.

But it still requires the AAC decoding & ac3 dynamic encoding to be decoded externally on my ONKYO receiver through s/w based ds filters or h/w based soundstorm(which I prefer).. non-pure-passthrough.

The soundstorm upsamples and encodes to 48K ac3.
And the general solution would be transcoding to 48K ac3 or DTS for the simple spdif passthrough until getting a AAC format supported receiver.

Rockaria · 4th August 2005, 03:00

I was able to soft ac3 encode->spdif passthrough using the ffdshow audio dsfilter alone with 48k resampling.

The problem was the ac3filter sticking just after the ffdshow audio when I set ac3filter priority higher in its option and output is set to ac3 in the ffdshow audio.
Obviously the ac3filter was trying to intercept the spdif-passthrough stream causing the hang up.

SeeMoreDigital · 4th August 2005, 09:10

Quote:

Originally Posted by Rockaria

Obviously the ac3filter was trying to intercept the spdif-passthrough stream causing the hang up.

Actually, you don't need to use AC3Filter at all

Rockaria · 4th August 2005, 09:34

Yeah, the key point(to use the ffdshow alone) is to set the ffdshow audio priority to higher level in the filter override(as E-male said) and setting the ac3filter's 'prefer other ds filter' checked(to exclude it from the player).

But because the current ffdshow ds filter version is having no DTS DRC processing, I am currently using both filters and either encoding(soundstorm or ac3filter).
The ac3filter(or possibly ffdshow audio filter) s/w ac3 encoding-passthrough sounded a bit better than the DICE.

21st February 2005, 20:45	#1 \| Link
oddball Registered User Join Date: Jan 2002 Posts: 1,101	AAC 5.1 to AC3 SPDIF passthru does not work. Audio stutters continuously. There appears to be a problem with libfaad2. Using Tremor on a Vorbis 5.1 file no problemo. Plays back as AC3 5.1 via SPDIF. But not AAC 5.1 Any ideas? Last edited by oddball; 21st February 2005 at 20:59.

3rd March 2005, 23:33	#10 \| Link
SeeMoreDigital Life's clearer in 4K UHD Join Date: Jun 2003 Location: Notts, UK Posts: 12,422	As I posted in another section of the forum, what I would like, is for FFdshow to offer a totally separate "AC3 Transcoding" application. So if there's anybody here who can whip up a very basic/no thrills AC3 transcoding application, that's very light on processing power... please do so! I thought maybe it could look/offer something like this: - Cheers __________________ \| I've been testing hardware media playback devices and software A/V encoders and decoders since 2001 \| My Network Layout & A/V Gear \|

6th March 2005, 23:22	#11 \| Link
oddball Registered User Join Date: Jan 2002 Posts: 1,101	AT present I have enabled ffdshow for AC3, DTS and MP3 with AC3 and DTS being passed to SPDIF and MP3 passing untouched as analog. I use AC3Filter for AAC and Ogg to 5.1 and 2.0 SPDIF. It kinda works. But ffdshow audio allowing to route each seperately would be ideal Especially allowing 2.0 or 1.0 MP3 Ogg and AAC to pass as analog whilst 5.1 AAC and Ogg get encoded to AC3 and sent to SPDIF. I still have the 44Khz AAC issue in ffdshow and have to use resampling if I want to use it. But at present I have just disabled it and let AC3Filter handle AAC instead as it handles 44Khz AAC with no problems whatsoever (For me). Last edited by oddball; 6th March 2005 at 23:29.

1st July 2005, 23:38	#12 \| Link
madman1980 Registered User Join Date: Apr 2005 Posts: 61	I'm trying to play a file now with ac3 5.1 48khz (dd) and my receiver only says it gets a dpl signal (stereo). How do I make ffdshow just use passthru? I don't want the signal to be messed with, just passed on to my receiver. I couldn't find any settings in ffdshow audio options that looked like it had anything to do with passthrough. Using ffdshow 19062005, letting ffdshow do the audio (ac3filter shouldnt be necessary anymore, right?). Last edited by madman1980; 1st July 2005 at 23:42.

4th August 2005, 09:34	#20 \| Link
Rockaria nobody's nobody Join Date: Mar 2005 Location: The Sun, somewhere around Posts: 553	Yeah, the key point(to use the ffdshow alone) is to set the ffdshow audio priority to higher level in the filter override(as E-male said) and setting the ac3filter's 'prefer other ds filter' checked(to exclude it from the player). But because the current ffdshow ds filter version is having no DTS DRC processing, I am currently using both filters and either encoding(soundstorm or ac3filter). The ac3filter(or possibly ffdshow audio filter) s/w ac3 encoding-passthrough sounded a bit better than the DICE. Last edited by Rockaria; 4th August 2005 at 09:37.

21st February 2005, 23:22	#4 \| Link
Ana Member Join Date: Nov 2003 Location: Finland Posts: 18	I had spdif passtrough problems with aac files too. My solution was resampling from 44.1khz to 48khz because my soundcard couldn't handle 44.1 khz via spdif properly. If I remember correctly nero makes only 44.1 khz aac-files...

21st February 2005, 23:25	#5 \| Link
oddball Registered User Join Date: Jan 2002 Posts: 1,101	That worked! You rule!

23rd February 2005, 11:40	#6 \| Link
Ana Member Join Date: Nov 2003 Location: Finland Posts: 18	Thanks. BTW there is a way to do 48kHz aac files with foobar using nero encoder. http://forum.doom9.org/showthread.ph...hreadid=67746&

24th February 2005, 06:12	#7 \| Link
oddball Registered User Join Date: Jan 2002 Posts: 1,101	If I was encoding them I would BTW AC3filter RC5 has no problem (Maybe it is resampling without showing it is).

3rd March 2005, 23:08	#9 \| Link
oddball Registered User Join Date: Jan 2002 Posts: 1,101	I use an M-Audio Revolution 7.1 soundcard.

3rd July 2005, 17:13	#13 \| Link
oddball Registered User Join Date: Jan 2002 Posts: 1,101	In the latest version there should be a drop down option next to the AC3 codec to select SPDIF.

3rd July 2005, 21:18	#14 \| Link
madman1980 Registered User Join Date: Apr 2005 Posts: 61	That worked. Thanks

6th July 2005, 05:18	#16 \| Link
E-Male mad computer-scientist Join Date: Mar 2002 Posts: 1,375	i have used the ffdshow(resampling) and ac3filter(encoding) combination successfully in mediaplayerclassic using the ffilter-overides options btw you can encode 48khz aac with besweet, just use commandline parameters with bsn (and NOT the config window)

6th July 2005, 09:45	#17 \| Link
Rockaria nobody's nobody Join Date: Mar 2005 Location: The Sun, somewhere around Posts: 553	Glad to know that the mediaplayer classic has the advanced filter option. I also was able to get 48K AACs in a step using foobar2k with a resampling plugin enabled. But it still requires the AAC decoding & ac3 dynamic encoding to be decoded externally on my ONKYO receiver through s/w based ds filters or h/w based soundstorm(which I prefer).. non-pure-passthrough. The soundstorm upsamples and encodes to 48K ac3. And the general solution would be transcoding to 48K ac3 or DTS for the simple spdif passthrough until getting a AAC format supported receiver.

4th August 2005, 03:00	#18 \| Link
Rockaria nobody's nobody Join Date: Mar 2005 Location: The Sun, somewhere around Posts: 553	I was able to soft ac3 encode->spdif passthrough using the ffdshow audio dsfilter alone with 48k resampling. The problem was the ac3filter sticking just after the ffdshow audio when I set ac3filter priority higher in its option and output is set to ac3 in the ffdshow audio. Obviously the ac3filter was trying to intercept the spdif-passthrough stream causing the hang up.