29 Nov
2010
29 Nov
'10
10:03 p.m.
Hi all,
I've got an ubuntu 10.04 system and I am using the sound card built
into my motherboard.
I would like to play some audio sampled at 96 kHz. But I don't appear
to have this enabled:
markrages@markrages-desktop:~$ cat /proc/asound/card*/codec#0 | grep rates
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
rates [0x60]: 44100 48000
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
But the same file lists "Codec: Analog Devices AD1986A". I looked up
the datasheet, and the AD1986A is able to support 96 kHz sampling with
"Double rate audio".
In audacity I generated a .wav file with a 24 kHz sine wave sampled at
96 kHz. It produces a flat line on the oscilloscope when I play it
back in audacity or sox. This makes me suspect that it's getting
downsampled to 48 kHz somewhere along the way. (I also tried playing
it back in alsaplayer and the result was an ugly on the 'scope. I
didn't try listening.).
I generated a 1 kHz wave which showed up loud and clear, so the audio
path is working.
(Before you ask, I'm not trying to play music for dogs, I'm trying to
generate FM stereo pilot tone.)
So, what do I need to do to get 96kHz playback working?
Regards,
Mark
markrages@gmail
--
Mark Rages, Engineer
Midwest Telecine LLC
markrages(a)midwesttelecine.com
29 Nov
29 Nov
11:03 p.m.
On Monday 29 November 2010 22:03:23 Mark Rages wrote:
Hi all,
I've got an ubuntu 10.04 system and I am using the sound card built
into my motherboard.
I would like to play some audio sampled at 96 kHz. But I don't appear
to have this enabled:
markrages@markrages-desktop:~$ cat /proc/asound/card*/codec#0 | grep rates
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
rates [0x60]: 44100 48000
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
But the same file lists "Codec: Analog Devices AD1986A". I looked up
the datasheet, and the AD1986A is able to support 96 kHz sampling with
"Double rate audio".
In audacity I generated a .wav file with a 24 kHz sine wave sampled at
96 kHz. It produces a flat line on the oscilloscope when I play it
back in audacity or sox. This makes me suspect that it's getting
downsampled to 48 kHz somewhere along the way. (I also tried playing
it back in alsaplayer and the result was an ugly on the 'scope. I
didn't try listening.).
I generated a 1 kHz wave which showed up loud and clear, so the audio
path is working.
(Before you ask, I'm not trying to play music for dogs, I'm trying to
generate FM stereo pilot tone.)
So, what do I need to do to get 96kHz playback working?
You need a soundcard that does 96kHz.
Intel hda is not suited for that. "High definition audio" is only suitable for
dvd and still is spec'ed for a lot of hizz and noise...
If its for a notebook your choice is somewhat limited compared to the choice
for pci/pcie capable machines. But there are several devices for low and high
prices out there that will do 96kHz and also 192kHz. Using a hda for this is
just asking for trouble and disappointment (if it even works).
Have fun,
Arnold
11:09 p.m.
On Mon, Nov 29, 2010 at 4:03 PM, Arnold Krille <arnold(a)arnoldarts.de> wrote:
You need a soundcard that does 96kHz.
Intel hda is not suited for that. "High definition audio" is only suitable for
dvd and still is spec'ed for a lot of hizz and noise...
If its for a notebook your choice is somewhat limited compared to the choice
for pci/pcie capable machines. But there are several devices for low and high
prices out there that will do 96kHz and also 192kHz. Using a hda for this is
just asking for trouble and disappointment (if it even works).
Soundcard snobbery doesn't answer my question. Let me rephrase:
- I have an AD1986A codec chip on my motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
Regards,
Mark
markrages@gmail
--
Mark Rages, Engineer
Midwest Telecine LLC
markrages(a)midwesttelecine.com
11:21 p.m.
On Mon, Nov 29, 2010 at 5:09 PM, Mark Rages <markrages(a)gmail.com> wrote:
Soundcard snobbery doesn't answer my question.
Let me rephrase:
he wasn't being snobbish. he might be wrong though.
- I have an AD1986A codec chip on my motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
because what the codec can do isn't the same as what the soundcard can do.
the use of the term "codec" in /proc/asound/card0/codec$0" is a bit
misleading.
11:29 p.m.
On Mon, Nov 29, 2010 at 04:09:25PM -0600, Mark Rages wrote:
- I have an AD1986A codec chip on my motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
Because:
* A particular HW design does not have to support everything
a chip could do.
* A driver does not necessarily support all hardware features.
And even if the HW and the driver would support 96 kHz, you
probably won't get 48 kHz bandwidth for the analog output.
You'll be lucky if you get 30 kHz or so.
Ciao,
--
FA
There are three of them, and Alleline.
11:32 p.m.
On Mon, Nov 29, 2010 at 4:21 PM, Paul Davis <paul(a)linuxaudiosystems.com> wrote:
On Mon, Nov 29, 2010 at 5:09 PM, Mark Rages
<markrages(a)gmail.com> wrote:
So lspci reports my soundcard as "nVidia Corporation MCP51 High
Definition Audio (rev a2)"
Looking around the web, I can see that others have 96 kHz sampling
available with this card, although with different (realtek) codecs.
Regards,
Mark
markrages@gmail
--
Mark Rages, Engineer
Midwest Telecine LLC
markrages(a)midwesttelecine.com
Soundcard snobbery doesn't answer my
question. Let me rephrase:
he wasn't being snobbish. he might be wrong though.
- I have an AD1986A codec chip on my
motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
because what the codec can do isn't the same as what the soundcard can do.
the use of the term "codec" in /proc/asound/card0/codec$0" is a bit
misleading.
11:33 p.m.
On Mon, Nov 29, 2010 at 4:29 PM, <fons(a)kokkinizita.net> wrote:
On Mon, Nov 29, 2010 at 04:09:25PM -0600, Mark Rages
wrote:
I'm using the Alsa driver as set up by Ubuntu. Would the OSS driver
be a better choice?
Regards,
Mark
markrages@gmail
--
Mark Rages, Engineer
Midwest Telecine LLC
markrages(a)midwesttelecine.com
- I have an AD1986A codec chip on my
motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
Because:
* A particular HW design does not have to support everything
a chip could do.
* A driver does not necessarily support all hardware features.
11:34 p.m.
On Mon, Nov 29, 2010 at 5:32 PM, Mark Rages <markrages(a)gmail.com> wrote:
So lspci reports my soundcard as "nVidia
Corporation MCP51 High
Definition Audio (rev a2)"
Looking around the web, I can see that others have 96 kHz sampling
available with this card, although with different (realtek) codecs.
the first thing to realize about the "Intel HDA" specification and all
devices that claim to use it is that its not really a specification at
all.
the second to realize about all devices that claim to use it is that
they are subject to completely arbitrary chip swaps, rewirings and
other "internal" changes that may or may not affect the extent to
which they support the non-specification.
11:36 p.m.
On Mon, Nov 29, 2010 at 5:33 PM, Mark Rages <markrages(a)gmail.com> wrote:
I'm using the Alsa driver as set up by Ubuntu.
Would the OSS driver
be a better choice?
That's a question that a number of us, myself included, would have to
plead the 5th Amendment on.
There is a school of thought which says that the OSS driver is NEVER,
EVER, EVER even a choice, let alone a better choice.
You might have a different experience, but you'll be outside the
operating bounds of most fo the people on this mailing list and the
overwhelming majority of other linux users.
--p
11:42 p.m.
On Mon, Nov 29, 2010 at 04:33:49PM -0600, Mark Rages wrote:
On Mon, Nov 29, 2010 at 4:29 PM,
<fons(a)kokkinizita.net> wrote:
No. To generate the analog signal to modulate an FM transmitter
for stereo you need at least 53 kHz usable bandwidth with a flat
amplitude respnse and perfect linear phase. The minimal practical
sample rate would be around 120 kHz. No audio interface I know of,
not even those work at 192 kHz, can do this.
You'd need an 'instrumentation/laboratory/scientific DA converter
for this to work at all.
Ciao,
--
FA
There are three of them, and Alleline.
On Mon, Nov 29, 2010 at 04:09:25PM -0600, Mark
Rages wrote:
I'm using the Alsa driver as set up by Ubuntu. Would the OSS driver
be a better choice? - I have an AD1986A codec chip on my
motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
Because:
* A particular HW design does not have to support everything
a chip could do.
* A driver does not necessarily support all hardware features.
11:55 p.m.
On Mon, Nov 29, 2010 at 4:41 PM, <fons(a)kokkinizita.net> wrote:
On Mon, Nov 29, 2010 at 04:33:49PM -0600, Mark Rages
wrote:
Right. I just want the pilot tone.
You are saying the 192 kHz soundcards roll off supersonic frequencies
between the DAC and the output jack? Or something else?
Regards,
Mark
markrages@gmail
--
Mark Rages, Engineer
Midwest Telecine LLC
markrages(a)midwesttelecine.com
On Mon, Nov 29, 2010 at 4:29 PM,
<fons(a)kokkinizita.net> wrote:
No. To generate the analog signal to modulate an FM transmitter
for stereo you need at least 53 kHz usable bandwidth with a flat
amplitude respnse and perfect linear phase. The minimal practical
sample rate would be around 120 kHz. No audio interface I know of,
not even those work at 192 kHz, can do this.
You'd need an 'instrumentation/laboratory/scientific DA converter
for this to work at all. On Mon, Nov 29, 2010 at 04:09:25PM -0600, Mark
Rages wrote:
I'm using the Alsa driver as set up by Ubuntu. Would the OSS driver
be a better choice? - I have an AD1986A codec chip on my
motherboard.
- According to its datasheet, AD1986A supports 96 kHz sample rate.
- /proc/asound/card0/codec#0 doesn't list 96 kHz.
Why?
Because:
* A particular HW design does not have to support everything
a chip could do.
* A driver does not necessarily support all hardware features.
30 Nov
30 Nov
12:33 a.m.
On Mon, Nov 29, 2010 at 04:55:12PM -0600, Mark Rages wrote:
That's 19 kHz, a sample rate of 48 kHz will do fine.
A residual of this shows up in every spectrum I take
from my FM receiver.
The only purpose of generating the 19 kHz pilot without
the actual difference signal would be to make the 'Stereo'
indication light up while the signal is actually mono :-)
No. To
generate the analog signal to modulate an FM transmitter
for stereo you need at least 53 kHz usable bandwidth with a flat
amplitude respnse and perfect linear phase. The minimal practical
sample rate would be around 120 kHz. No audio interface I know of,
not even those work at 192 kHz, can do this.
You'd need an 'instrumentation/laboratory/scientific DA converter
for this to work at all.
Right. I just want the pilot tone. You are saying the 192 kHz soundcards roll off
supersonic frequencies
between the DAC and the output jack? Or something else?
Most do, as they should. Any signal above 25 kHz or so is
just good to blow up your power amp.
Ciao,
--
FA
There are three of them, and Alleline.
6:52 a.m.
Paul Davis a écrit :
On Mon, Nov 29, 2010 at 5:33 PM, Mark Rages
<markrages(a)gmail.com> wrote:
Hum... Isn't that list the Linux Alsa Users list ? =:o))
I'm using the Alsa driver as set up by
Ubuntu. Would the OSS driver
be a better choice?
That's a question that a number of us, myself included, would have to
plead the 5th Amendment on.
There is a school of thought which says that the OSS driver is NEVER,
EVER, EVER even a choice, let alone a better choice.
You might have a different experience, but you'll be outside the
operating bounds of most fo the people on this mailing list and the
overwhelming majority of other linux users.
--p
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user(a)lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user
8:57 a.m.
Mark Rages wrote:
I would like to play some audio sampled at 96 kHz.
But I don't appear
to have this enabled:
markrages@markrages-desktop:~$ cat /proc/asound/card*/codec#0 | grep rates
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
rates [0x60]: 44100 48000
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
But the same file lists "Codec: Analog Devices AD1986A". I looked up
the datasheet, and the AD1986A is able to support 96 kHz sampling with
"Double rate audio".
"Double rate audio" is a special AC'97 mode that reuses some surround
channels to transport more samples for the two front channels. (AC'97
doesn't have enough bandwidth for more channels at 96 kHz.)
HDA assumes that all DACs can run at the same sample rate. However, it
looks as if this chip is just an AC'97 core that also has a HDA frontend,
so it cannot run all DACs at 96 kHz, so it completely disallows 96 kHz
when connected to a HDA controller.
(All HDA controllers would support 96 kHz playback, but as for AC'97,
only the Intel driver has double-rate support.)
(Before you ask, I'm not trying to play music for
dogs, I'm trying to
generate FM stereo pilot tone.)
So, what do I need to do to get 96kHz playback working?
A sound card that actually supports 96 kHz playback and that does not
filter out high frequencies in its analog output stage.
I've seen the M-Audio Audiophile 2496 advertized as being capable of
producing up to 48 kHz tones.
The Xonar D2 ran record frequencies of up to 90 kHz, but its output
filters out anything above 20 kHz. I've heard rumors that Asus changed
this in the later Xonar models after bat owners and the HiFi Voodoo
crowd complained; try the Xonar D1 or DX.
Regards,
Clemens
6:55 p.m.
On Tue, Nov 30, 2010 at 1:59 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote:
Mark Rages wrote:
Excellent answer. Thank you.
Regards,
Mark
markrages@gmail
--
Mark Rages, Engineer
Midwest Telecine LLC
markrages(a)midwesttelecine.com
I would like to play some audio sampled at 96
kHz. But I don't appear
to have this enabled:
markrages@markrages-desktop:~$ cat /proc/asound/card*/codec#0 | grep rates
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
rates [0x60]: 44100 48000
rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000
But the same file lists "Codec: Analog Devices AD1986A". I looked up
the datasheet, and the AD1986A is able to support 96 kHz sampling with
"Double rate audio".
"Double rate audio" is a special AC'97 mode that reuses some surround
channels to transport more samples for the two front channels. (AC'97
doesn't have enough bandwidth for more channels at 96 kHz.)
HDA assumes that all DACs can run at the same sample rate. However, it
looks as if this chip is just an AC'97 core that also has a HDA frontend,
so it cannot run all DACs at 96 kHz, so it completely disallows 96 kHz
when connected to a HDA controller.
(All HDA controllers would support 96 kHz playback, but as for AC'97,
only the Intel driver has double-rate support.)
4915
days inactive
4916
days old
linux-audio-user@lists.linuxaudio.org
14 comments
6 participants
participants (6)
-
Arnold Krille -
Clemens Ladisch -
fons＠kokkinizita.net -
fred -
Mark Rages -
Paul Davis