I've been having a lot of trouble with getting my Xonar DX to output DTS/Proper 5.1 to my receiver instead of PCM. I'm having trouble using both digital and analog connections. I think I've got it working, but I want to make sure I have it working correctly.
This is what it looks like using 6 Channel Direct/Analog mode on my Z5500 receiver:
http://i.imgur.com/JhnLoOX.png
Decoder info:
SpoilerInput format: SPDIF - 48000
User format: PCM Float - 0
Output format: PCM Float 5.1 48000
Filter chain:
(SPDIF - 48000) -> Demux -> (DTS 5.1 48000) -> AudioDecoder -> (Linear PCM 5.1 48000) -> AudioProcessor -> (PCM Float 5.1 48000) -> Dejitter -> (PCM Float 5.1 48000)
Demux
(SPDIF - 48000) -> FrameSplitter -> (SPDIF 5.1 48000) -> SPDIFParser -> (DTS 5.1 48000)
FrameSplitter
Stream format: SPDIF 5.1 48000
Bitstream type: 16bit low endian
Frame size: 2048
Samples: 512
Bitrate: 1536kbps
AudioDecoder
(DTS 5.1 48000) -> FrameSplitter -> DTSParser -> (Linear PCM 5.1 48000)
FrameSplitter
Stream format: DTS 5.1 48000
Bitstream type: 16bit big endian
Frame size: 2048
Samples: 512
Bitrate: 1536kbps
SPDIF stream type: 0xb
AudioProcessor
User format: PCM Float 5.1 48000
Dithering mode: auto (disabled)
Filter chain:
(Linear PCM 5.1 48000) -> Levels -> CacheFilter -> Resample -> Mixer -> BassRedir -> EqualizerMch/ConvolverMch -> DRC -> Dither -> AGC -> Delay -> CacheFilter -> Levels -> Converter -> (PCM Float 5.1 48000)
CacheFilter
Buffer size: 0ms (0 samples)
Resample
Soruce sample rate: 48000Hz
Sample rate: 48000Hz
Attenuation: 100.0dB
Quality: 1.0
Passthrough (no conversion)
Mixer
Input: L C R SL SR LF
Output: L C R SL SR LF
Buffered: false
Auto matrix: true
Normalize matrix: true
Vaoice control: true
Expand stereo: true
Center level: 0.0 dB
Surround level: 0.0 dB
LFE level: 0.0 dB
Gain: 0.0 dB
Input gains:
Output gains:
User matrix:
L C R SL SR LF CL CR BL BC BR
L: 1.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
C: 0.0 1.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
R: 0.0 0.0 1.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
SL: 0.0 0.0 0.0 1.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
SR: 0.0 0.0 0.0 0.0 1.0 0.0 0.0 0.0 0.0 0.0 0.0
LF: 0.0 0.0 0.0 0.0 0.0 1.0 0.0 0.0 0.0 0.0 0.0
CL: 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
CR: 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
BL: 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
BC: 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
BR: 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0 0.0
Resulting matrix:
L C R SL SR LF
L: 1.0 0.0 0.0 0.0 0.0 0.0
C: 0.0 1.0 0.0 0.0 0.0 0.0
R: 0.0 0.0 1.0 0.0 0.0 0.0
SL: 0.0 0.0 0.0 1.0 0.0 0.0
SR: 0.0 0.0 0.0 0.0 1.0 0.0
LF: 0.0 0.0 0.0 0.0 0.0 1.0
BassRedir
Enabled: true (active)
Crossover frequency: 80Hz
Bass destination: subwoofer
Bass gain: 0.0dB
EqualizerMch/ConvolverMch
Trivial processing (no convolution)
L: passthrough
C: passthrough
R: passthrough
SL: passthrough
SR: passthrough
LF: passthrough
DRC
Gain: 0.0
DRC: false
DRC power: 0.0
Loudness interval: 50ms (2400samples)
Attack: 100.0dB/s
Release: 50.0dB/s
Dither
Enabled: false
AGC
Gain: 0.0
Auto gain: true
Loudness interval: 50ms (2400samples)
Normalize: false
Attack: 100.0dB/s
Release: 50.0dB/s
Delay
Enabled: false
Units: samples
CacheFilter
Buffer size: 2775ms (133248 samples)
Converter
Output format: PCM Float
Channel order: L R C LF BL BR CL CR
Buffer size: 2048 samples
Dejitter
Enabled: true
Threshold: 100ms
Time shift: 0ms
Time scale: 1.0
It certainly sounds much better and more defined when I watched the same movie (World War Z) the other night. But shouldn't the "Input format:" be DTS? Is something wrong there? LAV audio decoder (Internal) says I'm "bitstreaming":
http://i.imgur.com/m18j2G8.png
I don't even know how I got LAV audio decoder doing anything in the first place though. I just uninstalled the k-lite pack because it was just giving me issues (DTS did not work at all with ffdshow and k-lite.. just static and a sped up picture). Do I need to get rid of LAV or AC3 to get proper DTS?
I got rid of LAV audio decoder by making AC3 the preferred decoder. Now it says the input is DTS 48khz, but everything else is the same. Shouldn't the output be DTS too? Why is it changing it to PCM?
Alternatively, I have an optical cable I can use which gives the same or similiar results (except the Output in AC3filter is SPDIF 48khz). Would that be better than analog? I'm assuming analog gives better quality and more control though (can only control volume with the receiver when using the optical cable).
Thanks in advance..!