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[maxxsid]

Hello,
I have a 4-year old Onkyo tx-sr703 receiver and it doesn't play nice with DTS/AC3 multichannel wav files. It plays them just fine but the playback is not gapless. When one track ends it switches to stereo (or some other default mode set in options) with quite loud and annoying relay-clicking sounds and then back to DTS or Dolby when the next track begins to play.
This is not an issue when it plays a movie DVD - then it clicks going from, say, the root menu to the main title and then there are no clicks for the duration of the movie. But when it's a DTS/AC3 audio album it's just unbearable. It pauses/clicks after every track.
(It would be interesting to hear if this is fixed in newer A/V receivers? As far as I got it from google, it's more or less the same for all receivers - just some of the receivers do this faster and not as loud)

My question here is:
-- why the CUE sheet files don't work with DirectShow decoding? If they worked that would fix my problem.  
Can this be implemented in MC? (i.e. cue sheets working for DirectShow decoding).

Thanks!
--max

from http://yabb.jriver.com/interact/index.php?topic=53878.0 :

The playback range values do not work through DirectShow decoding. (You are using the AC3 source and decoding filters for playing the DTS wav content, aren't you?)

You can convert the DTS encoded wav + cue albums file to individual DTS encoded track wav files. The DTS format will be preserved and the tracks will play gaplessly (I just tested it). MC must be temporarily set to use its standard wav decoding method in order to make the conversion work correctly.

[maxxsid]

Will these help the issue?

-- putting the wav's into mka container?
-- make a large multichannel flac with a cue sheet?

?

thanks!

[JimH]

This may be no help at all, but a couple of days ago I read a little of my receiver's manual.  It had several settings.  One was automatic, which it sounds like your receiver uses.  A couple of others allowed me to force a particular setting to always be in effect.  I don't remember more.  Sorry.

[maxxsid]

Yes... I googled and found the same info.
Yes, in my receiver everything was set to Auto or Last_Successfully_Used.
Unfortunately, the setup only allows to set either to DTS or to Dolby (AC3), so if I set it to DTS then AC3 stops playing... I'll mess with this some more... Maybe I'll tell AC3filter to covert everything to AC3 (which isn't nice quality-wise but at least there will be some sound. I hope).

Is there a chance to implement something about this on MC level?

Thanks!
 

This may be no help at all, but a couple of days ago I read a little of my receiver's manual.  It had several settings.  One was automatic, which it sounds like your receiver uses.  A couple of others allowed me to force a particular setting to always be in effect.  I don't remember more.  Sorry.

[JimH]

MC14.0.71 can output AC3.  That would be worth a try.  It's in Playback/DSP Studio/Output.

[maxxsid]

Haven't tried this yet.. but wouldn't that be "artificial" multichannel?
 

MC14.0.71 can output AC3.  That would be worth a try.  It's in Playback/DSP Studio/Output.

[maxxsid]

Well.. it's actually a bit different (and worse) for my receiver.
It only allows for Auto or DTS.
So, if I set it to Auto, it plays DTS _and_ AC3 but clicks between tracks; if I set it to DTS, it doesn't click between DTS tracks but it doesn't play AC3.
I guess, I'll just get used to the clicks or play via stand alone DVD player until J River offers some sort of a solution.

--max

Yes... I googled and found the same info.
Yes, in my receiver everything was set to Auto or Last_Successfully_Used.
Unfortunately, the setup only allows to set either to DTS or to Dolby (AC3), so if I set it to DTS then AC3 stops playing... I'll mess with this some more... Maybe I'll tell AC3filter to covert everything to AC3 (which isn't nice quality-wise but at least there will be some sound. I hope).

Is there a chance to implement something about this on MC level?

Thanks!
 

[Alex B]

Actually, JRiver added support for playback range to DirectShow audio files in the build 70. It makes possible to play wav + cue tracks through DirectShow decoding. Unfortunately, it doesn't seem to work gaplessly. After each track buffering happens after the track has ended and causes an additional pause (several seconds) even when the playback mode is set to gapless. I hope this is fixable.


It is still possible to use the oldest method and just play the wav files directly without using external DirectShow decoding, but to make that work the playback chain must be capable of bit perfect SPDIF output. The sound device must support ASIO or WASAPI in the exclusive mode (otherwise the Windows mixer destroys the DTS or DD code) and it must also support that when the file has the 44.1 KHz sample rate (many entry level SPDIF devices can play only 48 KHz). MC14 itself is bit perfect when all signal processing is disabled.

When the audio output is streamed bit perfectly and MC's internal playback engine is set to play gaplessly the receiver gets a continuous stream without any detectable pauses and it doesn't do any mode switching.

This has worked for years in several MC versions, but the sound device must be suitable for the task.

[maxxsid]

Actually, JRiver added support for playback range to DirectShow audio files in the build 70. It makes possible to play wav + cue tracks through DirectShow decoding. Unfortunately, it doesn't seem to work gaplessly. After each track buffering happens after the track has ended and causes an additional pause (several seconds) even when the playback mode is set to gapless. I hope this is fixable.

Great! Hope this gets polished eventually...

It is still possible to use the oldest method and just play the wav files directly without using external DirectShow decoding, but to make that work the playback chain must be capable of bit perfect SPDIF output. The sound device must support ASIO or WASAPI in the exclusive mode (otherwise the Windows mixer destroys the DTS or DD code) and it must also support that when the file has the 44.1 KHz sample rate (many entry level SPDIF devices can play only 48 KHz). MC14 itself is bit perfect when all signal processing is disabled.

I think that's what I've been doing...
I am using ASIO4All with the option "always Resample 44.1 <-> 48kHz" checked.
I do use AC3Filter with AC3/DTS passthrough to SPDIF. BTW, why to use it if it just passes through?
Should "the oldest method" work without AC3Filter?
I am using on-board sound (Realtek HD-Audio) with optical SPDIF-out. Tried an Asus Xonar D2X - didn't see any difference in the behavior.

[Alex B]

Should "the oldest method" work without AC3Filter?

It works and already worked before MC had any DirectShow audio options. I can just play my 44.1 KHz ripped DTS wav tracks with the standard playback engine without routing the output through DirectShow.

The playback chain:

DTS Wav file (Technically a standard stereo 16/44.1 audio file. The DTS signal is coded inside the stereo PCM audio stream)  >  MC's standard playback engine  (all DSP options disabled, internal volume at max, output set to ASIO)   >   ASIO driver (if ASIO4ALL, resampling must be disabled)  >  Terratec DMX 6fire 24/96 sound card  >  SPDIF output (optical or coaxial, both work)  >  Yamaha receiver (detects the DTS code in the signal and switches automatically to the DTS decoding mode)

I am using ASIO4All with the option "always Resample 44.1 <-> 48kHz" checked.

If you must do that in order to make standard 2-channel 44.1 kHz stereo files work through SPDIF then most likely your filters are decoding, resampling and re-encoding the 5.1 channel 44.1 KHz DTS tracks before they are send to the sound device and SPDIF.

[JimH]

If you are running Vista or Windows 7, try setting MC's Playback to WASAPI.

[maxxsid]

I am using XP and trying to use ASIO4ALL..

If you are running Vista or Windows 7, try setting MC's Playback to WASAPI.

[maxxsid]

Thank you so much Alex!
I am sorry, I am not very good at these things - still a bit confused.
Could you please answer a couple of more questions?

-- AC3Filter is not needed? Yes/No? I wonder why I started using it...
-- if AC3Filter is installed and active and set to passthrough AC3 and DTS and output set to "As Is" will it interfere with bit-perfect?
-- why it is recommended to use AC3Filter/ffdshow to play multichannel sound?

Thanks!

It works and already worked before MC had any DirectShow audio options. I can just play my 44.1 KHz ripped DTS wav tracks with the standard playback engine without routing the output through DirectShow.

The playback chain:

DTS Wav file (Technically a standard stereo 16/44.1 audio file. DTS signal is coded inside the stereo PCM audio stream)  >  MC's standard playback engine  (all DSP options disabled, internal volume at max, output set to ASIO)   >   ASIO driver (if ASIO4ALL, resampling must be disabled)  >  Terratec DMX 6fire 24/96 sound card  >  SPDIF output (optical or coaxial, both work)  >  Yamaha receiver (detects the DTS code in the signal and switches automatically to the DTS decoding mode)


If you must do that in order to make standard 2-channel 44.1 kHz stereo files work through SPDIF then most likely your filters are decoding, resampling and re-encoding the 5.1 channel 44.1 KHz DTS tracks before they are send to the sound device and SPDIF.

[Alex B]

-- AC3Filter is not needed? Yes/No? I wonder why I started using it...

It is not "needed" if you just want to play wave files that contain DTS (or DD) encoded PCM data - assuming your sound device (aka sound card) can output bit perfect data to SPDIF natively without resampling the output. If the manufacturer states that the device is ASIO compatible and provides an ASIO driver the device is more likely to be able to do that.

In any case you must first bypass the Windows mixer, which cannot be set to be fully bit perfect. Its mixer features are always available and like any high quality digital mixer it must add a very small amount of dithering noise in order to avoid audible distortion. This is desirable for standard audio playback, but it destroys any integrated DTS or DD code, which cannot be altered at all. ASIO (and WASAPI in the exclusive mode) can bypass the Windows mixer.

If the sound card has native support for the used sample rate and it can be set to output to SPDIF with its own control panel or if its SPDIF output is always active (like in my Terratec) it should be able to transmit unaltered audio signal.

if AC3Filter is installed and active and set to passthrough AC3 and DTS and output set to "As Is" will it interfere with bit-perfect?

It should be fine, but with 44.1 KHz files the sound card must support that sample rate natively. (AFAIK, all sound cards with SPDIF support 48 KHz without problems.)

-- why it is recommended to use AC3Filter/ffdshow to play multichannel sound?

It is necessary with video files and also with audio files when you want to decode the multichannel audio signal on PC and use an analog connection to the amp or powered speakers (i.e. connect the wires for 6 individual analog channels when the setup is 5.1.)

For DTS/DD wav audio files, SPDIF, and an external decoder/receiver the choice is yours. You may need to use the DirectShow playback system, SW decoding, resampling, and re-encoding if your HW does not support 44.1 KHz natively.

[maxxsid]

Thanks mucho, Alex!
It's all much clearer now.
Maybe I need to dig out my old but bit-perfect compatible CHAINTECH AV-710 card and experiment with it (I could successfully output multichannel sound with it back in MC11 days)

Thanks!

[JimH]

I am using ASIO4All with the option "always Resample 44.1 <-> 48kHz" checked.

Did you try with that option off?

[maxxsid]

Not yet but I will!
Now I have a feeling that I'll get it working..

Thanks for helping out!

Did you try with that option off?

[maxxsid]

Apparently, the option "always Resample 44.1 <-> 48kHz" in ASIO4ALL was the culprit...
Unchecked and all works without any hiccups now!
Thank you guys!

[Yaobing]

Actually, JRiver added support for playback range to DirectShow audio files in the build 70. It makes possible to play wav + cue tracks through DirectShow decoding. Unfortunately, it doesn't seem to work gaplessly. After each track buffering happens after the track has ended and causes an additional pause (several seconds) even when the playback mode is set to gapless. I hope this is fixable.

Does this happen only in cue tracks?  I threw a few wav tracks into Playing Now with Playback Range.  They played gaplessly.

[Alex B]

Does this happen only in cue tracks?  I threw a few wav tracks into Playing Now with Playback Range.  They played gaplessly.

How-to reproduce:

Set the wav file type to "JRiver audio engine (using  DS filters) and select the AC3File (source) and FDDShow (audio decoder) filters.

Download a free DTS wav sample file from this link:
http://www.diatonis.com/downloads/diatonis_dts_wav_secret-universe.zip (about 60 MB)

Use the attached cue sheet and play the tracks.