Mailing List Archive: MythTV: Mythtvnz

[OT] audio recording issue

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jhaar-ourshack-com at whanau

Jun 3, 2009, 10:03 PM

Post #1 of 7 (1254 views)
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[OT] audio recording issue

Hi there

Totally nothing to do with mythtv, but being "multimedia-philes" means
someone here may be able to answer my question. I've already submitted a
bugreport to Fedora (https://bugzilla.redhat.com/show_bug.cgi?id=489437)
but no-one appears interested.

Basically I find that ekiga and twinkle both have TERRIBLE recording
quality (tested uner FC8,F10 and F11). When I'm talking to someone over
them, the other end always comments on how bad the line sounds. However,
skype recording sounds perfect. Doing some digging shows me that using
the command-line ALSA tool "arecord file.wav" produces a similarly
crappy recording, but "arecord -f cd file.wav" produces a much better
sounding output - the big difference is the default is 8bit vs 16bit.

It smells to me like ekiga/twinkle aren't "telling" pulseaudio (although
I think this is an old issue and affects ALSA too) to choose a higher
quality recording format and that's the root cause of my problem?

Has anyone else experienced this, and is there a workaround? My
workstation and laptop both have Intel 82801G (ICH7 Family) soundcards,
and their audio-out has never been an issue - it's just the recording.
It maybe a issue with the card, but I'm not going to be able to swap out
the laptop that's for sure ;-)

Thanks

Jason

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jim at inode

Jun 3, 2009, 11:01 PM

Post #2 of 7 (1196 views)
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Re: [OT] audio recording issue [In reply to]

On Thu, Jun 4, 2009 at 5:03 PM, Jason Haar <jhaar-ourshack-com [at] whanau>
> It smells to me like ekiga/twinkle aren't "telling" pulseaudio (although
> I think this is an old issue and affects ALSA too) to choose a higher
> quality recording format and that's the root cause of my problem?

For Ekiga, does changing the order/selection of your audio codecs make
any difference?
(Edit -> Preferences ; Audio -> Codecs)

-jim

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jhaar-ourshack-com at whanau

Jun 4, 2009, 12:08 AM

Post #3 of 7 (1197 views)
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Re: [OT] audio recording issue [In reply to]

On 06/04/2009 06:01 PM, Jim Cheetham wrote:
>
> For Ekiga, does changing the order/selection of your audio codecs make
> any difference?
> (Edit -> Preferences ; Audio -> Codecs)
>
No - I tried all that. In the end I brought up an Asterisk server,
forced only G729 and G726 and still it's sounds awful - in fact I
basically don't hear any difference between the different codecs.
Fundamental issues like hearing static when I say "s" words is common.
Running "file" on the voicemails show they are all 8000Hz, whereas the
"asound -f cd" which sounds great is 44KHz - which is what I think the
problem is. However, I am not an audio-guru so it's complete guesswork
for me.

Jason

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hads at nice

Jun 4, 2009, 12:34 AM

Post #4 of 7 (1193 views)
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Re: [OT] audio recording issue [In reply to]

On Thu, 2009-06-04 at 19:08 +1200, Jason Haar wrote:
> No - I tried all that. In the end I brought up an Asterisk server,
> forced only G729 and G726 and still it's sounds awful - in fact I
> basically don't hear any difference between the different codecs.
> Fundamental issues like hearing static when I say "s" words is common.
> Running "file" on the voicemails show they are all 8000Hz, whereas the
> "asound -f cd" which sounds great is 44KHz - which is what I think the
> problem is. However, I am not an audio-guru so it's complete guesswork
> for me.

I'm not surprised, a lot of people find G729 and friends sound less than
good. Asterisk is narrowband (8 kHz) too, like the traditional phone
network.

If you're doing end to end VoIP and what is in between supports wideband
codecs (G722, CELT, some Speex etc.) for example FreeSWITCH then you can
get great sound. You'll need decent audio hardware too.

If you're using a VoIP provider or going out to the PSTN at all then it
will be narrowband.

Typically people find a hardware SIP phone gives much better call.
quality

hads
--
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier

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stephen_agent at jsw

Jun 4, 2009, 12:57 AM

Post #5 of 7 (1193 views)
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Re: [OT] audio recording issue [In reply to]

On Thu, 04 Jun 2009 19:34:27 +1200, you wrote:

>On Thu, 2009-06-04 at 19:08 +1200, Jason Haar wrote:
>> No - I tried all that. In the end I brought up an Asterisk server,
>> forced only G729 and G726 and still it's sounds awful - in fact I
>> basically don't hear any difference between the different codecs.
>> Fundamental issues like hearing static when I say "s" words is common.
>> Running "file" on the voicemails show they are all 8000Hz, whereas the
>> "asound -f cd" which sounds great is 44KHz - which is what I think the
>> problem is. However, I am not an audio-guru so it's complete guesswork
>> for me.
>
>I'm not surprised, a lot of people find G729 and friends sound less than
>good. Asterisk is narrowband (8 kHz) too, like the traditional phone
>network.

Yes, 8 kHz sampling is too little - sound will be distorted no matter
how good the codec is. The human voice does normally have frequencies
in it > 4 kHz that matter to voice quality.

>If you're doing end to end VoIP and what is in between supports wideband
>codecs (G722, CELT, some Speex etc.) for example FreeSWITCH then you can
>get great sound. You'll need decent audio hardware too.
>
>If you're using a VoIP provider or going out to the PSTN at all then it
>will be narrowband.

PSTN should be 64 kbit/s, which gives quite good quality with the
codecs normally used for that. I forget which ones they are. But
they are now used for all PSTN calls - the old landline phone is
connected to a card at the exchange that digitises the call. None of
the phone network is analogue now except the local loop.

>Typically people find a hardware SIP phone gives much better call.
>quality

I would have thought that using the same codec as is used in a
hardware SIP phone would give the same sound quality from a PC.

>hads

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hads at nice

Jun 4, 2009, 1:24 AM

Post #6 of 7 (1198 views)
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Re: [OT] audio recording issue [In reply to]

On Thu, 2009-06-04 at 19:57 +1200, Stephen Worthington wrote:
> PSTN should be 64 kbit/s, which gives quite good quality with the
> codecs normally used for that.

It's G711. The kbit/s rate isn't really that important when you're
talking about these things, more the sample rate (8k for G711).

It's okay quality, but not when you compare it to G722 and the other
wideband codecs, or CD quality etc.

> I would have thought that using the same codec as is used in a
> hardware SIP phone would give the same sound quality from a PC.

It possibly to get the same sound quality but that doesn't mean it
always happens, protocol implementations, processing and of course
acoustics all play a part.

Perhaps this would be better discussed on something such as the NZVoIP
group, it's really not at all on topic here.

hads
--
http://nicegear.co.nz
New Zealand's Open Source Hardware Supplier

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jhaar-ourshack-com at whanau

Jun 4, 2009, 1:53 AM

Post #7 of 7 (1194 views)
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Re: [OT] audio recording issue [In reply to]

On 06/04/2009 08:24 PM, Hadley Rich wrote:
>
>> I would have thought that using the same codec as is used in a
>> hardware SIP phone would give the same sound quality from a PC.
>>
>
> It possibly to get the same sound quality but that doesn't mean it
> always happens, protocol implementations, processing and of course
> acoustics all play a part.
>
>
As I mentioned - skype sounds great and so does "asound -f cd" - so it's
not a hardware problem. Also I'm using the same codecs our Cisco
softphones (Windows only) use and they sound fine. It seems to me that
recording voice is just not done in real numbers on Linux - and so it's
implemented poorly?

> Perhaps this would be better discussed on something such as the NZVoIP
> group, it's really not at all on topic here.
>
I'll take the hint. I'd certainly like to get it working. I'd actually
be able to get rid of the work phone I have at home if it worked well

Jason

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