Hi everyone!

Is anybody interested in SPDIF passthrough for AC3/DTS?

Here it is foo_spdif page

cool plugin, but it play my dts files to fast. i think this is the same problem like with an early version of Acropolis dts plugin.

Yes, well it's probably because your files are recorded at 44.1kHz and your sound card doesn't support spdif passthrough at such frequency. Unfortunately most sound boards have output fixed at 48kHz.

foobar2000 has a DSP called Resampler which should do the job.

I guess this doesn't work for Vista x64? All I get is: "Unable to open item for playback (Unsupported file format):[insert my dts or dd file here]. I tried both .AC3/.DTS and .SPDIF file extensions, same result.

I have no idea I assume your sound card driver for vista supports spdif playback, right?

If you wanna play *.ac3 and *.dts files, you must have the input plugins for those formats (foo_ac3 and foo_dts)

Didn't work here.

I tried with a clean install with/without foo_ac3 plugin

Without foo_ac3, foobar2000 crashed

With foo_ac3, it was normal playback, no spdif passtrough

Maybe that MSVCR80.dll library? Where needs to be added?

-Rozzo

Hi,

this kind of plugin is all that I need for happiness.

Unfortunately it doesn't work here.

foo_spdif Without foo-Ac3, foobar2000 crashes

foo-spdif With foo-Ac3 I get usual playback, not spdif

Added msv80 library into system32, no happiness

*.spdif appended to filenames, no happiness

Uh?
Rozzo

Ok, I'm working on few changes, please be patient

new release:
foo_spdif_v0.2

v0.2:
- fixed horrible bug preventing even a try at opening spdif for sample rates different than 48kHz
- optimized playback startup delay when forcing 48kHz
- added <FORCED!> tag in codec name for files forced to 48kHz
- added support for mpeg audio (for test purposes, works with ".spdif" extension only, any feedback is welcome)
- added support for stream type change, eg if there are dts and ac3 streams following each other in one file

additional requirements:
- foobar2000 v0.9.4 at least
- directX8 or newer

Grzesio,

Thank you very much for creating this plugin. It is the missing piece that I was looking for.

I tried v0.1b, and the only problem that I encountered is that it does not always set the sound card to the correct sampling rate. For example, when I play a 44.1kHz .AC3 file, sometimes it correctly sets the sound card to 44.1kHz, but sometimes it incorrectly sets the sound card to 48kHz. A similar problem happens when I play a 48kHz .AC3 file. In any case, the foobar2000 status bar always shows the correct sampling rate. It's just the sound card that is being set incorrectly sometimes. To get around the problem, I have to keep restarting the playback of the file until it eventually sets the correct sampling rate.

I will try v0.2 and report my results.

By the way, I am using the latest version of foobar2000 (v0.9.4.3), with the ASIO output plugin, on the M-Audio Delta DiO 2496 sound card (with the latest drivers), and Windows XP SP2.

Hi,

thanks for your work.

I don't know if I miss something:

When I play my AC3 files with creative mediasource (windows XP, Creative X-Fi, digital bitstream output activated), I get a pop-up "Dolby Digital Bitstream Out" and the display on my Denon AVR 1706 Receiver reads "Dolby Digital EX". In this condition, music sounds perfect through my six speakers, loud, clear and integrated, sooo satisfying. In this mode, I can control music's volume only through the receiver controls.

Same effect obtained with Zoom Player and Ciberlink Power DVD.

With Foobar and spdif plugin (I tried Direct Sound, Kernel Streaming and Asio aoutput) I get only standard Playback, Receiver applies its PLII or NEO 5x1 effects, so no way to hear my AC3 as they.

My question is if this spdif passtrough plugin can send the ac3 files directly to the receiver as Dolby Digital bitstream from Foobar or I'am misunderstanding something.

For me is very important because I'm attached to foobar2000 and want to use it as my standard player, but the hearing experience with Dolby Digital Bitstream Output for AC3 files is far superior to the 5x1 plugin solutions I find otherwise. It's real surround music!!

Thanks for the info, Erik. I'm waiting for your feedback on v0.2.

Can anyone confirm that mpeg audio (mp1,mp2,mp3) passthrough works or not?

I tried v0.2, and 44.1kHz .AC3 files have no problem now. Thanks for correcting it.

However, 48kHz .AC3 files still have the same problem as before. I will try to explain it in more detail. The problem happens when you use any of the following methods to start playing the .AC3 file:

1. Press the Play button.
2. Double-click it in the playlist.
3. Press the Next button.
4. Press the Previous button.

In addition, the behavior depends on whether the sound card is set to 44.1kHz or 48kHz before you start playing it.

If the sound card is set to 44.1kHz before you start playing it, then foobar2000 sets the card to 48kHz, which is the correct sampling rate. However, the progress slider and the elapsed time in the status bar will get stuck after 1 second. foobar2000 continues to play the file anyway. When the end of the file is reached, foobar2000 will not automatically play the next file in the playlist. It just stops completely.

Or, if the sound card is set to 48kHz before you start playing it, then foobar2000 sets the card to 44.1kHz, which is the incorrect sampling rate. However, there is an easy way to work around this situation. While it is playing, you can simply press Pause twice (or press Pause and then Play), which causes foobar2000 to set the card to 48kHz, which is the correct sampling rate. In this case, the progress slider and the elapsed time in the status bar will not get stuck. foobar2000 will automatically play the remaining files in the playlist as well, and it will play them with no problems.

I figured out that the best way to start playing a 48kHz .AC3 file is to perform these three steps:
1. Start playing it.
2. If it is playing at 48kHz, then start playing it again, so it is playing at 44.1kHz now, even though it is incorrect.
3. Press Pause twice, so it is playing at 48kHz now, which is correct.

By the way, I also tried playing MPEG Audio files by renaming them to .SPDIF. It seems that foobar2000 is sending audio to the sound card at the correct sampling rate, but the external decoders do not seem to be receiving or recognizing the audio.

Ops, that is very very odd what you have discovered
Thanks for detailed explanation of the problem. I will try to do something about it.
To be shure, does it mean that if you don't click anything (just let the playlist play itself continously) then everything is ok?
Does the problem become only when you are changing from one playing track to another or is it present also when you are starting to play from silence (stopped state)? Could you test it again with output plugin set to null renderer?
And also I would like to know if the <FORCED!> tag is visible at any time in the status bar.

About mpeg audio, does it mean that your receiver is capable of decoding mpeg audio? Because mine simply is not

Everything is OK if I just let the playlist continue automatically from file to file. I can click on the progress slider, and everything is still OK.

The problem happens in both of those cases.

I will test that.

I never saw the tag, but I wasn't always looking for it.

I have connected to two devices: a Denon AVR-3806 receiver and a JVC SU-DH1 headphone adapter. I am not sure if the Denon supports any MPEG audio format. The owner's manual states: "MUSE 3.1 and MPEG multi-channel audio are not available to North American consumers – same is true for Dolby’s AAC." In addition, the serial protocol manual states that MPEG-2 AAC is "Invalid at AVR model." It is not clear if this means that the Denon does not support MPEG audio at all, or just does not support MPEG-2 AAC in particular. As for the JVC, the owner's manual states that it supports MPEG-2 AAC, up to 5.1 channel. It has an LED indicator labeled "MPEG-2 AAC" that is supposed to light up, but I have not seen it light up at all. Anyway, it seems that they are not receiving any audio, since nothing lights up, and there is just silence.

I tested Null Output. The progress slider and elapsed time never get stuck, and it seems to be playing at the correct sampling rate, or at least the total elapsed time seems to be accurate.

So it looks like some kind of incompatibility with your asio output but I have also found some weak points in the source code that may cause your problem. I going to rewrite part of the plugin but unfortunately it may take a little longer.

To be more specific, the problem happens when I play padded and unpadded 48kHz AC3 files with these extensions: .AC3, .AC3WAV, .SPDIF, and .WAVAC3.

(Padded files are files that have been padded with zeros in order to match the size of a PCM stream.)

The problem does not happen when I play padded .WAV files. For example, I can simply change the extension of a padded file from .AC3WAV to .WAV, and the problem goes away. In fact, I can play padded .WAV files without even using foo_spdif, since foobar2000 just passes it through as if it were PCM audio.

I also tested 48kHz DTS files. Same problem.

44.1kHz DTS files have no problems though.

I hope foo_spdif_v0.3 will help on that.

foobar 0.9.4f + foo_spdif_v0.3 + foo_input_dts 0.1.7
dts file 44100 works fine, but i had to rename my .dts files to .spdif

thanks for plugin !

Sorry Rozzo, I missed your post.

For me it looks like the spdif plugin doesn't work at all. Did you try to rename your ac3 files with .spdif extension?
By the way, you are right, spdif plugin sends ac3 files directly to your external receiver without decompressing it. It doesn't even use foobar's output plugin but its own. You can switch foobar's output to "null output" to test if spdif plugin really do its job.
And of course, your receiver should indicate Dolby Digital ® logo.

v0.3 works great for me. I no longer have problems playing 48kHz AC3/DTS files.

Thank you!!!


By the way, you mentioned this on your site:
- dropped support for mpeg audio since the main idea is to pass-through multi-channel audio

Some MPEG audio formats support multi-channel, such as MPEG-2 AAC. I think it would be useful if foo_spdif could pass through those multi-channel formats to the receiver (even though some receivers cannot decode them).

OK, I should have been more specific about that. By "mpeg audio" I meant only mp1,mp2,mp3 files but AAC is quite different thing to pass through spdif and as far as I know AAC is now defined as MPEG-4 standard.
I agree that it would be nice to include AAC support but unfortunately I couldn't get right specification so far, that is something like IEC 61937-6. I also have no equipment to test it

Works fine for me but just until it meets with local-codepage (cyrillic) characters in path.
For example:
Unable to open item for playback (Could not open file.):
"D:\Моя музыка\Album1\CDImage.dts"

PS: Anyway, thanks for this cool plugin.

Hi Grzesio,

happiness here,
plugin started to work when instead of cheking all the three options in the options page, I checked only ac3>spdif.
Back to Foobar, thanks a lot,
Rozzo.

Thanks for the plugin it's something i've been needing for a while,

but it doesn't work . I have Vista x64, with a realtek ac97 chipset.

I believe it's in the 600 series.

Anyhow I only get two channel audio output. So it seems as the Foo_AC3 and Foo_Input_DTS seem to decoding the file and ignoring the Foo_SPDIF

May I play multi channel WavePack file with this plugin ?

I have some .wv files that plays fine on Foobar (latest), but if I rename it to .SPDIF it says "Unable to open item for playback (unknow file format or corrupt data)"


(XP Pro SP2)

Ah I got it to work in Vista x64 but only at 48khz, at 44.1kHz it's a no go for me. Both 44.1kHz DTS and DD files are not passed through correctly (my receiver flickers between DD/DTS and PCM).

I am in audio creation mode and have bit-perfect playback enable so I should be able to passthrough 44.1kHz material (I could in XP x64). Although it might be a Vista X-Fi driver bug (I don't have another program that supports passthrough of 44.1kHz material atm, but I will setup MPC soon and report back). Either way, keep up the good work.

EDIT: Same thing happens on MPC and AC3Filter with 44.1kHz DD/DTS files, so I guess it's a X-Fi bug.

I'm going to correct it in the nearest possible future. I had to drop the project for a while but I will be back

Unfortunately, no.

Hi Grzesio,
I get this error:

My audio codec is Realtek ALC850 (integrated on Asus A8N32-SLI mobo), latest Vista driver, "DS: Realtek Digital Output" chosen as output device.
Any idea?

I get this message also, but using windows xp x64 and a different soundcard.
I tried all devices in the list (I have bitperfect playback on this soundcard with KS 32-bit setting but that didn't work either)

Switch off analog output.

I have same gigbyte mobo with ALC850. Divx+ ac3 works fine.
But some problem dts. Foobar + foo_spdif play dts on yamha 459 (detect dts 5.1) , but time on foobar hold on. Always 0:00. One song ending, next never start. Until i push button start. I have foobar2000_0.9.4.3 + foo_spdif 0.3
With foo_input_dts, time on foobar is ok, but only 2 channel.
After i close foobar and open it, always msg:"Module uninstalled: Module foo_spdif.dll has been removed since last run. Would you like to keep its settings?". Then i must instal foobar again...

ociec

What is your os?
Could you try to switch your foobar output to "Null output" and check if it helps with unfreezing the progress bar?

Vista Ultimate PL

"Null output" - Now foobar play ok. Progres bar counting time. Songs change to next. thanks.


I uninstal foobar, delete folder and instal again. Foobar dont show this msg.

ociec

I'm glad to hear that.
I think there is still some problem with spdif pass-through in Vista sound drivers
The Null output is for test purposes only. Spdif will work fine but other media won't be audible.
You can try different output plugins and find out one working properly, if any.

latest stablest version:
foo_spdif_v1.0

Thanks for the component!
Although, when I send the signal through spdif, the soundcard shows a popup indicating that its decoding the signal. But when I play files using foo_input_dts, it doesn't, so I guess the decoding is being done in software (But it sounds 5.1 so there is no problem). Using only the spdif component, I can't get playback of the files. But having both of them activated (foo_input_dts & foo_spdif) I can hear DTS as 5.1 and the soundcard pop-up shows, so the decoding is being done there, everything is ok. The only problem are the APEv2 tags that the foo_input_dts allows me to put on the .dts files, embedded cuesheets and that stuff. It doesn't get recognized (so I think it's only spdif or software decoding for now). Is there a way to make both of these great plugins live together?
Thanks in advance.

Hi Everyone,I can't browse the site above with Firefox or IE.
Can anybody tell me how can I get the plugins? thanks!

I think the name of spdif plugin is a little confusing and suggest that this is pure output plugin. In fact, it is specific combination of input-output plugins You can have only one input plugin for any media file type (only one will work) if you want foo_input_dts then you should disable dts support in foo_spdif plugin (DTS->SPDIF option in foobar preferences), otherwise, you have to remove foo_input_dts.
I have no plans for APEv2 tags support for now.



Can anyone confirm this problem? It works fine for me with ie6 and opera.

Hi Grezsio,

after some time of using the plugin I found uncomfortable changing the extension of ac3 files to .spdif

No way to solve this inconvenient?, say open ac3 files directly in foobar2000 without intermediate operations?

ys,
Rozzo

Thanks for the answer!

Works fine in Maxthon & IE7!

Cannot connect to d/l page too by IE7 & Firefox. Could someone reply by attached file or u/l to somewhere? Thanks.

Thanks for this GREAT plugin!
Unfortunately the SPDIF output of my sound card is not recognized by the plugin (I use a Terratec Aureon 5.1 USB MKII). This is a common pb with a lot of sound cards, which can easily be solved by an option like "Output SPDIF as PCM" used by AC3Filter (cf. specifically here).

Is there any hope that such an option will be included? Thanks!

you don't have to change the extension to .spdif if you remove other input plugins for ac3 files.

Well, frankly, sending ac3 as pcm could be very annoying and even dangerous not only for the eqipment but also for your ears I know you may hate me but I think that buying a new card would be a better choice.

We all live in Taiwan.