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WASAPI misses start of track if rate/depth different to previous track, [moved as not a 3rd Party Plugin]
extracampine
post Dec 18 2012, 13:47
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I'm having a problem with music playback in Foobar when using the WASAPI plugin.

If I'm using WASAPI (event), then when I select a track of a different bitrate to the playing track, playback doesn't start at the beginning - it starts around 5 or 10 seconds into the track. This is independent of whether I had stopped the previously playing track first.

It only happens when the bitrate of the tracks changes, i.e. if I select another 16bit/44000 Hz track it doesn't happen, but if I then select a 24bit/96000Hz track it does.

Interestingly, when I use the WASAPI (push) setting, it only happens when I am going UP in bitrate, i.e. selecting a higher bitrate track after playing a lower bitrate tracks. It does not happen the other way round.

All my music is in FLAC format. Any help appreciated!!
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Peter
post Dec 18 2012, 14:48
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It would help if you could tell us your exact configuration (soundcard model, soundcard driver version, Windows version, WASAPI plug-in version).
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extracampine
post Dec 18 2012, 15:09
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Certainly sir -

Soundcard: none
Windows: Windows 7 Ultimate SP1
WASAPI plug-in version: 3.1 (the latest as far as I understand)

Output in Foobar is listed as "WASAPI (push) : Speakers (Weiss Firewire IO Audio)". When setting up the DAC I had to install some firewire driver thing also.

There is no soundcard as audio is sent digitally via a firewire card (VIA card) to an external DAC (Weiss DAC2). The DAC2 is set to auto-sample rate switching ("Allow WDM to change sample rate").

Thanks
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Peter
post Dec 21 2012, 12:07
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Thanks for the report. I'll see what I can do about it, though it looks like a device-specific problem. One solution would be to resample everything to the same sample rate to prevent device reinitialization which seems to be problematic.
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extracampine
post Apr 12 2013, 17:05
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I'm still getting issues with this. Am I the only one? Would really appreciate some help with it!
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SoNic67
post Apr 13 2013, 20:02
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Your DAC switches clock between 44.1 and 48 (multiples). It takes some time.
If you use DirectSound, Windows does re-sample everything to multiples of 48kHz, that's why you don't hear that pause. Maybe you want to use a resampler in the chain - I use the SoX based one for my E-MU card (that has the same setup of two separated clocks).
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extracampine
post Apr 27 2013, 19:54
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QUOTE (SoNic67 @ Apr 13 2013, 19:02) *
Your DAC switches clock between 44.1 and 48 (multiples). It takes some time.
If you use DirectSound, Windows does re-sample everything to multiples of 48kHz, that's why you don't hear that pause. Maybe you want to use a resampler in the chain - I use the SoX based one for my E-MU card (that has the same setup of two separated clocks).


I don't think it's the DAC time, as when the track starts playing it's further forward than the time spent waiting!
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terrorindeed
post May 11 2013, 08:40
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I am having a similar problem with the wasapi (Push or Event) plugin. The problem exists over the HDMI (Intel HD Audio) connection, optical (Realtek HD)playback is fine.
When I select a FLAC track, the track begins playing however the audio doesn't kick in for approximately 1.5 seconds. If I switch tracks manually either of the same bitrate/Sampling or a different one, same thing, no audio for 1.5 seconds.

Alternatively, if I create a playlist of varying bitrate/sampling, the first track manually selected behaves same as above. Yet, if I then leave the playlist to play through each track in order, there is no audio problem.

I have played around with the delay buffer settings to no avail. I would gladly use the optical output and avoid this mess altogether, but I use the HDMI connection for DTS-HD Master Audio.

I have a onkyo tx NR-609 and onkyo HT R-391 and have tried the HDMI connection to my Samsung 27inch monitor as well.
The front display on both Onkyo AVR's re-initializes when the described problem happens. And I therefore see this as the plugin telling the AVR something incorrectly, resulting in the re-initialization. (As If I'm switching between listening mode or even inputs).

I have not used any previous Wasapi versions as I have just recently moved into FLAC /lossless audio, previously being an iTunes user sad.gif

Thanks!

Windows 7 SP1 64bit Professional
Gigabyte z77-up5th
i7-3770K
32GB DDR 1600 Ram
Intel HD Audio Driver (Current)
Wasapi 3.2.3

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Sandrine
post May 11 2013, 11:35
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Have you tried foo_dsp_pregap to see if you can initialize your receiver silently?
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Propheticus
post May 11 2013, 14:57
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Isn't this inherent to the way many digital a/v connections operate? (and thus outside the capabilities of Foobar2000 to solve?)
As far as my understanding of the HDMI protocol goes it operates a little like this: When opening a stream the source negotiates with the destination device to determine it's capabilities and to set the samplerate/bit-depth that will be used. This is done before any actual audio/video data is transmitted. Every time the source format changes the stream has to be stopped and started again to negotiate a new setting for the stream. This might be the case for this firewire connection too..
This renegotiating (and/or clock setting change in the destination device) could be causing the silence you're talking about.

A simple solution has already been given by Peter: resample to the same rate/depth

edit: found a forumpost talking about the same issue, pointing to the possibility it's due to the DAC and the fact it's buffered: http://forums.naimaudio.com/topic/buffered...switching-issue

This post has been edited by Propheticus: May 11 2013, 15:06
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terrorindeed
post May 11 2013, 17:10
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Thanks for quick replies everyone!

Sandrine- The DSP pregap has reduced the silence by approximately 1250ms, even if I configure the pregap to create a 1250-4000ms the slight gap remains. Could this be the Onkyo's DAC limitations?

Propheticus- I'm just now reading into the suggested post, thanks! My issue is not bitrate/sampling dependent though, it is manual track selection dependent. Resampling unfortunately fixes nothing for me. If I select a 16bit 44.1KHz track I see the AVR re-initialize, if I then jump manually to another 16bit/44.1KHz track same thing, further, a 24bit 96Khz track with the produce the same silence/reinitialization even with the above mentioned DSP pregap. Argh. lol.

I can't thank you guys enough for the quick and precise assistance!
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Propheticus
post May 11 2013, 22:34
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You could try preventing the stopping and starting of the stream between tracks by using a crossfader. By default most don't crossfade on manual change, but can be set to do so.
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terrorindeed
post May 12 2013, 06:33
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I'll definitely try the crossfade tomorrow. I spent most of the afternoon setting up plex HT and have discovered that the problem does not exist, using the built in wasapi event. No re-initialization on same/varying sampling/bitrate, no problem, weird.
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