ALSA Dmix and Tempo Changes.
#1
I'm attempting to use ALSA dmix plugin to solve the lack of the Dual Audio output in Eden. I also use it so that numerous other programs can use output at once. Sometimes I just want to fire up mplayer from ssh to play a stream rather than do anything GUI, I also run Shairport instead of XBMC's implementation of AirPlay (because I want to! And that's my choice!)

So I came up with a very very simple asound.conf that I thought would fix everything.

Code:
pcm.!default {
    type plug
    slave.pcm "dmixer"
}

pcm.dmixer  {
    type dmix
    ipc_key 1024
    slave {
        pcm "hw:0,1"
    }
        slave {
                pcm "hw:0,0"
        }
    bindings {
        0 0
        1 1
    }
}

ctl.dmixer {
    type hw
    card 0
}

And it works great. I even fired up 2 instances of shairport and used my iPhone and iTunes to be a fake DJ. It works great... except with XBMC.

It took me an entire movie to figure it out and it wasn't until I started watching CSI, but sound is just a slight bit slower tempo. As far as I can tell XBMC is playing everything at 1.0x speed but all of the women sound a bit like men and all the men like Barry Manilow.

Does anyone have any ideas?
Code:
GRANT ALL PRIVILEGES ON `xbmc_%`.* TO 'xbmc'@'%';
IF you have a mysql problem, find one of the 4 dozen threads already open.
Reply
#2
sounds to me like the audio sampling rate is running at something like 48KHz but the actual audio info is sampled at 44.1KHz, or something to that effect. Do you know any of the above, i.e what the audio rate is in the show you watched? Also, what are you using to connect to your audio out from the box, is it over hdmi or not?
Reply
#3
boblablah Wrote:sounds to me like the audio sampling rate is running at something like 48KHz but the actual audio info is sampled at 44.1KHz, or something to that effect. Do you know any of the above, i.e what the audio rate is in the show you watched? Also, what are you using to connect to your audio out from the box, is it over hdmi or not?

Coax SPDIF to a receiver.

Code:
Requested audio codec family [mpg123] (afm=mpg123) not available.
Enable it at compilation.
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
AUDIO: 48000 Hz, 2 ch, floatle, 128.0 kbit/4.17% (ratio: 16000->384000)
Selected audio codec: [ffmp3float] afm: ffmpeg (FFmpeg MPEG layer-3 audio)

Without any changes to alsa elsewhere (All I did was change to plughw:0,1 in XBMC) it works just fine.
Code:
GRANT ALL PRIVILEGES ON `xbmc_%`.* TO 'xbmc'@'%';
IF you have a mysql problem, find one of the 4 dozen threads already open.
Reply

Logout Mark Read Team Forum Stats Members Help
ALSA Dmix and Tempo Changes.0