24 Aug
2014
24 Aug
'14
1:08 p.m.
I have a USB DAC that can only handle 16/44.1 as input and output. I
think ALSA will resample everything to 16/44.1 automatically, but I'd
like that to happen with the highest quality resampler which I think
is samplerate_best. I use xfce4 and I don't want to install
pulseaudio. I've added the following to /etc/asound.conf:
defaults.pcm.rate_converter "samplerate_best"
Should that do it? How can I verify that it's working?
- Grant
24 Aug
24 Aug
2:05 p.m.
Grant wrote:
I've added the following to /etc/asound.conf:
defaults.pcm.rate_converter "samplerate_best"
Should that do it?
Yes. (If that converter is installed.)
How can I verify that it's working?
Run aplay with -v:
$ aplay -v something.wav
Playing WAVE 'something.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S32_LE)
Converter: libspeex (builtin)
Protocol version: 10002
Its setup is:
...
Regards,
Clemens
5:38 p.m.
I've added
the following to /etc/asound.conf:
defaults.pcm.rate_converter "samplerate_best"
Should that do it?
Yes. (If that converter is installed.)
How can I verify that it's working?
Run aplay with -v:
$ aplay -v something.wav
Playing WAVE 'something.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S32_LE)
Converter: libspeex (builtin)
Protocol version: 10002
Its setup is:
... 
2:45 p.m.
On Sun, 24 Aug 2014, Grant wrote:
I have a USB DAC that can only handle 16/44.1 as input
and output. I
think ALSA will resample everything to 16/44.1 automatically, but I'd
Normally, the application connecting to ALSA looks at the port to find out
what sample rates it can do and adjusts accordingly. Any recording
application should lock to the interface rate with no resampleing. MP3s,
Oggs and other compressed/lossy formats do internal resampling/filtering
to match the desired output sampling rate anyway, but most of them are
44.1k to begin with. Wav files and flac and other no lossy formates are
the only ones where resampling is needed if they are not already 44.1k. In
general any wav files will be ones you recorded and already be the right sample
rate.
I think what I am saying is that for most cases the sample rate of your
audio IF doesn't matter. So adding resampling to everything doesn't make
sense... maybe try without first.
like that to happen with the highest quality resampler
which I think
is samplerate_best. I use xfce4 and I don't want to install
pulseaudio. I've added the following to /etc/asound.conf:
defaults.pcm.rate_converter "samplerate_best"
Should that do it? How can I verify that it's working?
Playback a 48K wav file. if there are no extra clicks or pops it is fine.
--
Len Ovens
www.ovenwerks.net
5:48 p.m.
I have a USB
DAC that can only handle 16/44.1 as input and output. I
think ALSA will resample everything to 16/44.1 automatically, but I'd
Normally, the application connecting to ALSA looks at the port to find out
what sample rates it can do and adjusts accordingly. Any recording
application should lock to the interface rate with no resampleing. MP3s,
Oggs and other compressed/lossy formats do internal resampling/filtering to
match the desired output sampling rate anyway, but most of them are 44.1k to
begin with. Wav files and flac and other no lossy formates are the only ones
where resampling is needed if they are not already 44.1k. In general any wav
files will be ones you recorded and already be the right sample rate.
I think what I am saying is that for most cases the sample rate of your
audio IF doesn't matter. So adding resampling to everything doesn't make
sense... maybe try without first.
26 Aug
26 Aug
12:45 p.m.
I have a USB DAC that can only handle 16/44.1 as input
and output. I
think ALSA will resample everything to 16/44.1 automatically, but I'd
I think what I am saying is that for most cases the sample rate of your
audio IF doesn't matter. So adding resampling to everything doesn't make
sense... maybe try without first. 
27 Aug
27 Aug
5:02 p.m.
On Tue, Aug 26, 2014 at 8:45 AM, Grant <emailgrant(a)gmail.com> wrote:
I think you're saying that the ALSA resampler won't be used if the
upstream application does the resampling itself. Is that correct?
How can I find out if ALSA is the one resampling in a particular
scenario?
Is there any way to find out if this is happening? Do some
applications really do this?
depends on the device the application opens.
If you tell it to open a device with a name like "hw:N" or
"hw:CARDNAME"
then ALSA will offer only the capabilities present in the audio hardware.
If you open it with a name like "plughw:N" or "plughw:CARD" then ALSA
will
do everything it can to honor the format, sampling rate etc. asked for by
the application (including resampling, channel multiplexing etc.).
I have a USB
DAC that can only handle 16/44.1 as input and output. I
think ALSA will resample everything to 16/44.1 automatically, but I'd
I think what I am saying is that for most cases the sample rate of your
audio IF doesn't matter. So adding resampling to everything doesn't make
sense... maybe try without first. 
25 Aug
25 Aug
1:57 a.m.
Date: Sun, 24 Aug 2014 07:43:53 -0700
From: len(a)ovenwerks.net
To: emailgrant(a)gmail.com
CC: linux-audio-user(a)lists.linuxaudio.org
Subject: Re: [LAU] ALSA: always use samplerate_best
MP3s,
Oggs and other compressed/lossy formats do internal
resampling/filtering
to match the desired output sampling rate anyway
What are you going on about? Data files do not do any internal resampling!!
On that note I've noticed my 48kHz mp3s do not play correctly on Rhythmbox, which uses
GStreamer, but are fine in Audacious which doesn't. Rhythmbox is using the GStreamer
Jack plugin as I have fully disabled PA though and wonder if this may be the issue...
Anyway don't mean to derail the thread. ;)
Dale.
3563
days inactive
3566
days old
linux-audio-user@lists.linuxaudio.org
8 comments
5 participants
participants (5)
-
Clemens Ladisch -
Grant -
Kaza Kore -
Len Ovens -
Paul Davis