Hi,
Motorola h700.
I have plugz installed, and the corresponding asound.conf from it.
fc6 and headset paired.
I tell new alpha skype 1.4 to use headset for in, out and ring(as
headsetd is running)
When i try test call, it says problem with audio playback? However,
the headset beeps at this time, so there's definitely some kind of
communication.
What to do? Am I supposed to be using hijacker?
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Will 2.6.21 final still require the patch?
On 5/16/07, Robert Huitl <[email protected]> wrote:
> On Mittwoch, 16. Mai 2007, John H. wrote:
> > Did you have to use the kernel patch? I am trying to get h700 working
> > with skype without patching kernel.
>
> I did not test again without the patch. But it's strongly recommended to have
> it applied for using SCO.
>
> Regards,
> Robert
>
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On Mittwoch, 16. Mai 2007, John H. wrote:
> Did you have to use the kernel patch? I am trying to get h700 working
> with skype without patching kernel.
I did not test again without the patch. But it's strongly recommended to have
it applied for using SCO.
Regards,
Robert
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Did you have to use the kernel patch? I am trying to get h700 working
with skype without patching kernel.
On 5/15/07, Robert Huitl <[email protected]> wrote:
> On Dienstag, 15. Mai 2007, you wrote:
> > On Tuesday 15 May 2007 21:22, Robert Huitl wrote:
> > > Okay after some debugging I know what the problem is.
> > > 2.) Offering more sampling rates than 8 kHz would be nice to have, but
> > > I'm not familiar with the design philosophies around ALSA. It's probably
> > > not a good idea to implement resampling in each and every sound plugin
> > > like pcm_sco. Does ALSA have some resampling facilities that can be
> > > plugged in between an application and a plugin? Maybe some magic lines in
> > > .asoundrc?
> >
> > I do this with a2dp, I'm sure the same could apply to sco.
> > I use an .asoundrc file which looks like this:
>
> Thanks a lot for this configuration. I got Skype running with it, even though
> it's freezing almost always when I call the echo test service (and both input
> and output are set to headset-sco-resample). But in a few cases it actually
> worked very nicely.
>
> However, if only input _or_ output is set to the headset device, it's quite
> stable. So recording is fine, playback is fine, but both of them at a time
> freeze Skype.
>
> For reference, this is my .asoundrc:
>
> pcm.headset-sco-resample {
> type plug
> slave {
> pcm "headset-sco"
> rate 8000
> format S16_LE
> channels 1
> }
> }
> pcm.headset-sco {
> @args [BDADDR TIMEOUT]
> @args.BDADDR {
> type string
> default "xx:xx:xx:xx:xx:xx" # Put your HS address here
> }
> @args.TIMEOUT {
> type integer
> default 6000
> }
> type sco
> bdaddr $BDADDR
> timeout $TIMEOUT
> }
>
> Might be some problem with ALSA, though.
>
> Regards,
> Robert
>
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On Dienstag, 15. Mai 2007, you wrote:
> On Tuesday 15 May 2007 21:22, Robert Huitl wrote:
> > Okay after some debugging I know what the problem is.
> > 2.) Offering more sampling rates than 8 kHz would be nice to have, but
> > I'm not familiar with the design philosophies around ALSA. It's probably
> > not a good idea to implement resampling in each and every sound plugin
> > like pcm_sco. Does ALSA have some resampling facilities that can be
> > plugged in between an application and a plugin? Maybe some magic lines in
> > .asoundrc?
>
> I do this with a2dp, I'm sure the same could apply to sco.
> I use an .asoundrc file which looks like this:
Thanks a lot for this configuration. I got Skype running with it, even though
it's freezing almost always when I call the echo test service (and both input
and output are set to headset-sco-resample). But in a few cases it actually
worked very nicely.
However, if only input _or_ output is set to the headset device, it's quite
stable. So recording is fine, playback is fine, but both of them at a time
freeze Skype.
For reference, this is my .asoundrc:
pcm.headset-sco-resample {
type plug
slave {
pcm "headset-sco"
rate 8000
format S16_LE
channels 1
}
}
pcm.headset-sco {
@args [BDADDR TIMEOUT]
@args.BDADDR {
type string
default "xx:xx:xx:xx:xx:xx" # Put your HS address here
}
@args.TIMEOUT {
type integer
default 6000
}
type sco
bdaddr $BDADDR
timeout $TIMEOUT
}
Might be some problem with ALSA, though.
Regards,
Robert
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On Tuesday 15 May 2007 21:22, Robert Huitl wrote:
> Okay after some debugging I know what the problem is.
> 2.) Offering more sampling rates than 8 kHz would be nice to have, but I'm
> not familiar with the design philosophies around ALSA. It's probably not a
> good idea to implement resampling in each and every sound plugin like
> pcm_sco. Does ALSA have some resampling facilities that can be plugged in
> between an application and a plugin? Maybe some magic lines in .asoundrc?
>
I do this with a2dp, I'm sure the same could apply to sco.
I use an .asoundrc file which looks like this:
pcm.a2dpd {
=A0=A0=A0=A0=A0=A0=A0=A0type a2dpd
}
pcm.rs0 {
=A0=A0=A0=A0=A0=A0=A0=A0type plug
=A0=A0=A0=A0=A0=A0=A0=A0slave {
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0pcm "a2dpd"
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0rate 44100
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0format S16_LE
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0channels 2
=A0=A0=A0=A0=A0=A0=A0=A0}
}
pcm.!default {
=A0=A0=A0=A0=A0=A0=A0=A0type plug
=A0=A0=A0=A0=A0=A0=A0=A0slave.pcm "rs0"
}
----
It will resample the audio given to the a2dp plugin so it is always 44.1kHz.
Cheers,
Tim
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Okay after some debugging I know what the problem is.
On Dienstag, 15. Mai 2007, Robert Huitl wrote:
> 1) Skype tries to use a sample format that is not supported at all or
Skype uses the sample format SND_PCM_FORMAT_S16_LE which is supported by the
pcm_sco plugin, but it also requests a sampling rate of 48 KHz, which is not.
Having patched alsa-lib to trace the calls to snd_pcm_hw_params_set_rate*(), I
get the following output when I try to initiate a call:
DEBUG: snd_pcm_hw_params_set_rate_near(48000)
DEBUG: snd_pcm_hw_params_set_rate(48000)
[pause]
DEBUG: snd_pcm_hw_params_set_rate(48000)
[error message appears]
When I change SCO_RATE in pcm_sco.c from 8000 to 48000 and set the headset as
the input device only, Skype goes past the "problem with audio" message (the
recorded sound is garbage, of course).
So to conclude,
1.) 48 kHz is, to say the least, a rather high value for sampling from a
microphone. This is probably not intended by Skype and might just very well
be a bug as it's an alpha version.
2.) Offering more sampling rates than 8 kHz would be nice to have, but I'm not
familiar with the design philosophies around ALSA. It's probably not a good
idea to implement resampling in each and every sound plugin like pcm_sco.
Does ALSA have some resampling facilities that can be plugged in between an
application and a plugin? Maybe some magic lines in .asoundrc?
Regards,
Robert
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On Dienstag, 15. Mai 2007, Robert Huitl wrote:
> So either
> 1) Skype tries to use a sample format that is not supported at all or
> 2) the sample format is okay but it fails because of the missing kernel
> patch 3) there's some other problem ;-)
>
> You could try the patch to see if it helps.
Okay I just tried the patch myself. arecord works fine with the s16_le sample
format, but Skype does not.
Robert
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On Montag, 14. Mai 2007, John H. wrote:
> Motorola h700.
Jabra BT.620S here
> I have plugz installed, and the corresponding asound.conf from it.
>
> fc6 and headset paired.
>
> I tell new alpha skype 1.4 to use headset for in, out and ring(as
> headsetd is running)
>
> When i try test call, it says problem with audio playback? However,
> the headset beeps at this time, so there's definitely some kind of
> communication.
I can confirm this problem with Skype 1.4 Build 58. I tried setting only one
of the devices to use headsetd, but it works for neither playback nor
capturing.
I noticed that Skype seems to probe all devices when I initiate a call,
because headsetd establishes a connection even though all three (input,
output, ringing) devices are set to "default" (i.e. the internal sound card).
> What to do? Am I supposed to be using hijacker?
I tried to get headsetd working with other programs, for example arecord. This
might point to a problem with headsetd not supporting the sample format Skype
requests:
$ arecord -Dheadset-sco
Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono
arecord: set_params:900: Sample format non available
Trying a different sample format reminds me of a kernel patch you can find on
this list. It is required for reliable SCO communication:
$ arecord -Dheadset-sco -f s16_le
Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono
ALSA lib pcm_sco.c:277:(sco_headset_hw_params) Unable to set number of SCO
buffers : please upgrade your Kernel !
arecord: set_params:961: Unable to install hw params:
[...]
In both cases arecord fails to use the headsetd sound device.
So either
1) Skype tries to use a sample format that is not supported at all or
2) the sample format is okay but it fails because of the missing kernel patch
3) there's some other problem ;-)
You could try the patch to see if it helps.
Regards,
Robert
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Here's the output of headsetd, which "connects" after i hit button on
headset, but the rest is what outputs when I try to use skype with it.
headsetd -n
headsetd[20485]: Bluetooth headset daemon version 0.5
headsetd[20485]: Incoming RFCOMM hs connection from 00:16:8F:F9:97:BB accepted
headsetd[20485]: Changing state: Idle-->Connected
headsetd[20485]: Changing state: Connected-->Ready
headsetd[20485]: Changing state: Ready-->Opening
headsetd[20485]: SCO channel opened handle=0x0031 mtu=64
headsetd[20485]: Changing state: Opening-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Changing state: Zombie-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Changing state: Zombie-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Changing state: Zombie-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Changing state: Zombie-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Changing state: Zombie-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Changing state: Zombie-->Streaming
headsetd[20485]: Appli closed socket
headsetd[20485]: Changing state: Streaming-->Zombie
headsetd[20485]: Nobody uses SCO channel anymore, closing it.
headsetd[20485]: Changing state: Zombie-->Connected
On 5/14/07, John H. <[email protected]> wrote:
> Hi,
>
> Motorola h700.
>
> I have plugz installed, and the corresponding asound.conf from it.
>
> fc6 and headset paired.
>
> I tell new alpha skype 1.4 to use headset for in, out and ring(as
> headsetd is running)
>
> When i try test call, it says problem with audio playback? However,
> the headset beeps at this time, so there's definitely some kind of
> communication.
>
> What to do? Am I supposed to be using hijacker?
>
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