Re: [Linphone-users] up sample from 8000hz to 48000hz possible ?

[Simon Morlat]

Le mercredi 7 juin 2006 15:16, mcquaid mcquaid a écrit :
> I am interested in this as well, as I have the same issue with rates.  I've
> tried with dmix but it doesn't seem to upsample properly.
>
> I was curious why you said 1.3.99.4 was necessary if it's an issue only
> solved by alsa.
I just meant that with current stable branch it's difficult to use default 
device since you need to add a line manually in the config file...
With 1.3.99 you just need to click on the alsa device you need.
I don't have problems with dmix with my intel8x0 chipset.

>
> My other issue with sound is I want to use dmix for output but not input as
> this causes breakups on a lot of soundcards mic in.  Is there a way to
> specify hw for input?
Yes of course.

>
> Also, in trying to solve this rate issue other ways, can one configure
> linphone to use esd or jack?  I know normally, I wouldn't want to introduce
> more latency, but if one of those upsample properly I'd take the delay.  Is
> there current documentation on the config file?
You can use artsd (need arstc devel file to build linphone with artsd 
support). But personnaly I prefer dmix/dsnoop.

I had to manually change my asoundrc to tune some parameters (otherwise there 
were really alsa problems):

# Set default sound card
# Useful so that all settings can be changed to a different card here.
pcm.snd_card {
     type hw
     card 0
}

# Allow mixing of multiple output streams to this device
pcm.dmixed {
        type dmix
        ipc_key 1024
        slave {
            pcm "snd_card"
            period_time 0
            period_size 2048
            buffer_size 4096
            rate 44100
        }
        slave.pcm "snd_card"
}

# Allow reading from the default device.
# Also known as record or capture.
pcm.dsnooped {
        type dsnoop
        ipc_key 2048
        slave{
                pcm "snd_card"
                channels 2
                period_size 1024
                buffer_size 4096
                rate 44100
                periods 0
                period_time 0
        }
}

# This is what we want as our default device
# a fully duplex (read/write) audio device.
pcm.duplex {
     type asym
     playback.pcm "dmixed"
     capture.pcm "dsnooped"
}

###################
# CONVERSION PLUG #
###################
# Setting the default pcm device allows the conversion
# rate to be selected on the fly.
# duplex mode allows any alsa enabled app to read/write
# to the dmix plug (Fixes a problem with wine).
pcm.!default {
     type plug
     slave.pcm "duplex"
}

pcm.dsp0 {
        type plug
        slave.pcm "asymed"
}



Simon


>
> On 5/29/06, Simon Morlat <address@hidden> wrote:
> > Hello,
> >
> > Is there an alsa driver for this audio chip ? If any, maybe the alsa
> > library
> > can do the trans-rating in software, especially I you configure
> > your .asoundrc to use dmix/dsnoop plugins.
> > You'll have to use 1.3.99.4 prerelease at
> > http://download.savannah.nongnu.org/releases/linphone/unstable/source/
> >
> > Simon
> >
> > Le Vendredi 26 Mai 2006 22:42, gary ng a écrit:
> > > Hi,
> > >
> > > I have a crippled audio chip which only does 48000Hz,
> > > is it possible/easy to up sample the incoming voice
> > > stream from 8000Hz to it ?
> > >
> > > I tried to use esd daemon but all I got is garbage. If
> > > directly through /dev/dsp, I hear nothing.
> > >
> > > thanks for any info/hint/help in advance.
> > >
> > > BTW, how can I specify linphonec to use jack instead
> > > of /dev/dsp ? I failed to find any documentation about
> > > the syntax to be used for this in the rc file.
> > >
> > > regards,
> > >
> > > gary
> > >
> > >
> > >
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