Official OPPO BDP-103 Owner's Thread - Page 815 - AVS Forum | Home Theater Discussions And Reviews
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post #24421 of 24430 Old 02-18-2017, 04:48 PM
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Originally Posted by Ephi82 View Post
I have been using my 103 with some success to pull audio files off my lap top and play them over the wi Fi network in my house.

I had done the same with my 93.

I understand that the 103 supports the ability to push files at the 103 from say my laptop?

Does anyone have experience in how to do this?
If you install Twonky ($20 for the license) on your laptop as a DLNA server, it has the ability to push files to a renderer like the 103.

However, I prefer to use my Android phone as the controller, with Twonky as the file server and BubbleUPnP Server - which is free - acting as an intermediary making the Twonky library available remotely from your phone over the net with Android BubbleUPnP ($5 at the Google Play Store) and when your phone is on your wifi network, that same Android app can push music and movies from your laptop to the 103 and any other renderers you have.

I set BubbleUPnP to send all files at full resolution locally but as 128k mp3s over the net to my phone when listening remotely, to avoid running up my phone bill.

You can use the FooUPnP plugin for Foobar2000 as both a file server syndicated by BubbleUPnP Server and as a client/renderer/controller like the android app - so I run that as a client on my upstairs PC, with its sound card plugged into my hifi, letting me play music in bed using my phone as the controller, and on my laptop to listen to music from home when traveling. In the car, I run a cord from the phone's earphone jack into my car stereo's line input!

The advantage of Twonky over FooUPnP as the file server is that Twonky shows you embedded album art while browsing as well as while playing, while FooUPnP only shows it while playing. Also, Twonky can be set to auto update its index when you add new stuff, while FooUPnP has to be manually updated, including closing and reopening Foobar2000.

Both the Android BubbleUPnP and FooUPnP (which are written by the same programmer) let you do word searches through your library to find things.


Last edited by Philnick; 02-18-2017 at 04:52 PM.
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post #24422 of 24430 Old Yesterday, 09:54 AM
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post #24423 of 24430 Old Yesterday, 01:59 PM
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Question Album Art Playing (SACD) dsf files?

I've ripped a number of SACDs to dsf using my 103. I've then taken the dsf files and updated tags and added album art. When I playback the dsf files from my PC to my 103 over my LAN, the tags are correct but no album art is displayed. Album art displays fine for MP3s and FLAC files (over the LAN).

Has anyone been able to display album art when playing dsf files through their 103?

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post #24424 of 24430 Old Today, 07:00 AM
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Album art doesn't display for .dsf files on the Oppo.

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post #24425 of 24430 Old Today, 09:02 AM
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Album art doesn't display for .dsf files on the Oppo.
Thanks for confirming that. Kind of a shame that it doesn't. Also, it would be nice if SACD album art was available through Gracenote. I'll keep adding album art to my dsf files in the hope that some day that Oppo supports it.

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post #24426 of 24430 Old Today, 09:20 AM
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Originally Posted by kokishin View Post
Thanks for confirming that. Kind of a shame that it doesn't. Also, it would be nice if SACD album art was available through Gracenote. I'll keep adding album art to my dsf files in the hope that some day that Oppo supports it.
The path of least resistance is to just convert the files to 24/88 FLAC. You won't hear any difference, and its likely that your processor does the same thing anyway. (it has to in order to apply any processing)
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post #24427 of 24430 Old Today, 09:29 AM
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Originally Posted by rdgrimes View Post
The path of least resistance is to just convert the files to 24/88 FLAC. You won't hear any difference, and its likely that your processor does the same thing anyway. (it has to in order to apply any processing)
I do that with all my multichannel SACD rips. They are all getting fed through my Pioneer SC-87 processing anyway and all that processing is PCM - and in the end flac files are just a ton more convenient.
It takes some extra time (DSD2Flac on Mac or DBPowerAmp Music Converter both do a good job), but it sounds great.

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post #24428 of 24430 Old Today, 02:49 PM
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Originally Posted by rdgrimes View Post
The path of least resistance is to just convert the files to 24/88 FLAC. You won't hear any difference, and its likely that your processor does the same thing anyway. (it has to in order to apply any processing)
I took your advice and converted a stereo dsf file to 24/88 flac using foobar2000. Seems ok.

I recall reading somewhere that there may be some issue converting a multichannel dsf file to flac with respect to the .1 LFE channel. Something to do with the LFE channel requiring +10dB compensation adjustment. Please advise.

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post #24429 of 24430 Old Today, 03:53 PM
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Quote:
Originally Posted by kokishin View Post
I took your advice and converted a stereo dsf file to 24/88 flac using foobar2000. Seems ok.

I recall reading somewhere that there may be some issue converting a multichannel dsf file to flac with respect to the .1 LFE channel. Something to do with the LFE channel requiring +10dB compensation adjustment. Please advise.
That's up to your processor. Most will do the 10db boost to any LFE channel input. Shouldn't be too difficult to hear one way or the other. But that boost happens in the processor.
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post #24430 of 24430 Old Today, 04:41 PM
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Quote:
Originally Posted by kokishin View Post
I took your advice and converted a stereo dsf file to 24/88 flac using foobar2000. Seems ok.

I recall reading somewhere that there may be some issue converting a multichannel dsf file to flac with respect to the .1 LFE channel. Something to do with the LFE channel requiring +10dB compensation adjustment. Please advise.
Sony's design for SACD uses the LFE channel differently from everybody else.

Everybody else uses the LFE channel as a place to hold *LOUD* bass. So the content in that channel is recorded -10dB down and has to be boosted +10dB on playback to restore it to the level it should be at compared to the other speaker channels. AVRs, for example, almost universally add +10dB boost to the channel by default.

But SACD (as Sony designed it) does not do that. SACD records LFE without that -10dB attenuation. So if +10dB boost is applied during listening that channel will be too hot.

Now, when playing DSD digital audio content, whatever is processing that can make a guess (almost certainly correct) that this came from an SACD and apply -10dB attenuation to that LFE channel so that a subsequent, default, +10dB boost will leave things "correct".

But when you convert the DSD digital audio to another digital audio format, like LPCM, that provenance is lost. So the AVR, for example can't know not to provide the +10dB boost.

Ideally then, whatever is doing the CONVERSION from DSD to something else should apply -10dB attenuation to the LFE channels AS PART OF that conversion. In your case, that means the tool you are using to produce those FLAC files.

You'll have to ask people familiar with the tool whether it does that by default, or can be set to do that.

----------------------------------

This is such an ungodly can of worms that studios have taken countermeasures. Many Studios author their 5.1 SACD content so that there IS NO AUDIO in the .1 channel. I.e, it is really 5.0 authoring packaged as 5.1. *ALL* the bass is authored into the main speaker channels.

Some older SACDs, on the other hand, took the approach that, Sony notwithstanding, they were going to author 5.1 SACD content including the -10dB attenuation of the .1 channel as part of the authoring. I.e., the same thing EVERY OTHER FORMAT was doing with that .1 channel. These are, of course, truly problematic recordings today since the fixes built into the conversion algorithms in modern gear will now attenuate that channel yet ANOTHER -10dB.

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