From: Takashi Iwai <tiwai@suse.de>
To: Daniel James <daniel@64studio.com>
Cc: alsa-devel@lists.sourceforge.net, Free Ekanayaka <free@64studio.com>
Subject: Re: Card won't accept new sample rate?
Date: Tue, 29 Aug 2006 16:40:42 +0200 [thread overview]
Message-ID: <s5h3bbf61w5.wl%tiwai@suse.de> (raw)
In-Reply-To: <44F44FEA.1040301@64studio.com>
At Tue, 29 Aug 2006 15:32:10 +0100,
Daniel James wrote:
>
> Hi Takashi,
>
> > OSS does software sample rate conversion as default.
>
> Right, that figures.
>
> > Check your /proc/asound/card0/codec97#0/ac97#0-0 file, whether it
> > contains "VRA" in Extended ID section. If not, it's 48k-only codec.
>
> Indeed it does not - thanks for the clue :-)
>
> I have put an extra soundcard in one of the test machines showing the
> problem. The new card has the ice1712 chipset, and this changes sample
> rate perfectly.
>
> Is there any change that has been made to ALSA recently, which would
> mean that a 48k-only chipset no longer does sample rate conversion in
> software by default? If not, could this be something broken in Debian's
> configuration of ALSA?
If you start JACK, usually it prefers "hw" PCM, which means the direct
access to the hardware without resampling, up/downmix, and format
conversion (also without soft-mixing). The "default" PCM is with
everything, i.e. dmix, and plug layer.
So, I guess this is intentional behavior.
Takashi
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next prev parent reply other threads:[~2006-08-29 14:40 UTC|newest]
Thread overview: 5+ messages / expand[flat|nested] mbox.gz Atom feed top
2006-08-29 13:19 Card won't accept new sample rate? Daniel James
2006-08-29 13:29 ` Takashi Iwai
2006-08-29 14:32 ` Daniel James
2006-08-29 14:40 ` Takashi Iwai [this message]
2006-08-29 15:00 ` Daniel James
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