XBMC-12 beta 3: two sound problems
#1
Hi,

I'm getting some sound problems with all Frodo betas. I'm using amd64 build on Nvidia Ion box with NVidia [HDA NVidia], ALC887 soundcard connected by TOSlink to an external amp. Now, I have to choose as audio device "ALC887 Digital" while previously (on version 11) it worked also on default.

1. I can't get <streamsilence>0</streamsilence> to work. From what I understood it should even be default behaviour. However, XBMC blocks audio device all the time. It doesn't happen when I chose device to be "analog".
2. I lost bit-perfect playback. I have an easy and extremally rigorous test method - a DTS file disguised as a .wav. If audioplayer sends it perfectly to the hardware decoder, then it picks it up as DTS and decodes it. However, if any change in the stream occured (samplerate, amplification, sample format conversion), I only hear static. With version 11 I could play those files, but now I can't. I tried enabling <audiophile>1</audiophile> but that didn't help either.

Apart from those problems, everything works fine.
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#2
Hi tomaszg. I just tested a dts-wav (Engima MCMCX AD DTS 5.1 surround encoded in a wav) and it played back perfectly, with the amp in my signature showing DTS and 5.1 channels. I am using HDMI with an ATI 6450HD GPU.

Can you post a Debug Log?

There were issues with suspending ALSA output to release the audio device, so I doubt that feature is working on ALSA setups right now until we can do more work there. That issue won't be solved for Frodo despite some efforts by some of our Linux devs.
System: XBMC HTPC with HDMI WASAPI & AudioEngine - Denon  AVR-3808CI  - Denon DVD-5900 Universal Player  - Denon DCM-27 CD-Changer
- Sony BDP-S580 Blu-Ray  - X-Box 360  - Android tablet wireless remote - 7.1 Streem/Axiom/Velodyne Surround System
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#3
Here's the log: http://pastebin.com/QXyHLRFb

I now see what the problem is: my dts-wav is (what I forgot to mention) compressed by FLAC. That shouldn't make any difference, but it does. For some reason XBMC changes format sample to AE_FMT_FLOA

Code:
13:17:14 T:3010087744    INFO: CSoftAE::LoadSettings - Audiophile switch enabled
13:17:14 T:3010087744    INFO: CSoftAE::InternalOpenSink - keeping old sink with : AE_FMT_FLOAT, FL,FR, 44100hz
13:17:14 T:3010087744   DEBUG: CSoftAE::InternalOpenSink - Using speaker layout: 2.0
13:17:14 T:3010087744   DEBUG: CSoftAE::InternalOpenSink - Internal Buffer Size: 4096
13:17:14 T:3010087744   DEBUG: AERemap: Downmix normalization is enabled
13:17:14 T:3010087744   DEBUG: CSoftAEStream::CSoftAEStream - Converting from AE_FMT_S16NE to AE_FMT_FLOAT

It is completely unnecessary, since my soundcard doesn't support FLOAT in this setting:
Code:
13:16:54 T:3012512448    INFO: CAESinkALSA::InitializeHW - Your hardware does not support AE_FMT_FLOAT, trying other formats
13:16:54 T:3012512448    INFO: CAESinkALSA::InitializeHW - Using data format AE_FMT_S32NE

However, when I extract flac to plain wav file, XBMC plays it nicely. I suppose there is some mess in the code with the choice of prefered sample format, if a lossless format is subjected to two sample format conversions before playing it Smile Of course, my hardware supports AE_FMT_S16NE.
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XBMC-12 beta 3: two sound problems0