[linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
9 Sep
2004
9 Sep
'04
3:58 p.m.
Hi,
Can someone suggest what tools I would use to accomplish converting
two mono 24-bit, 96KHz files into a single 96KHz stereo wave file? Would
sndfile-convert be able to do this? sox? If you have it handy, what
command would you suggest?
Just for kicks I imported the mono files to Ardour tracks and then
cross wired them to a stereo track where I am recording them, but this
seems like a sill way to accomplish a task like this. My bad ability at
the command line.
After I've done that I'll likely try going through Jamin at 96K,
bouncing the output, and then I'll need to resample to 44.1K and
downsize to 16-bit. How are people here doing dither on Linux? What sort
of results are you getting?
Thanks in advance for any inputs.
Cheers,
Mark
9 Sep
9 Sep
4:19 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, Sep 09, 2004 at 10:41:50AM -0700, Mark Knecht wrote:
Hi,
Can someone suggest what tools I would use to accomplish converting
two mono 24-bit, 96KHz files into a single 96KHz stereo wave file? Would
sndfile-convert be able to do this? sox? If you have it handy, what
command would you suggest?
Just for kicks I imported the mono files to Ardour tracks and then
cross wired them to a stereo track where I am recording them, but this
seems like a sill way to accomplish a task like this. My bad ability at
the command line.
After I've done that I'll likely try going through Jamin at 96K,
bouncing the output, and then I'll need to resample to 44.1K and
downsize to 16-bit. How are people here doing dither on Linux? What sort
of results are you getting?
Thanks in advance for any inputs.
ecasound!
ecasound -a:1 -i:mono1.wav \
-a:2 -i:mono2.wav \
-a:1,2 -f:24,2,96000 -o:stereo.wav \
-a:2 -erc:1,2 -eac:0,1
Do that all on one line. The returns are escaped to make it easier to
read. This creates 2 chains, one for each input mono wav file. Both
chains are connected to the same output file. You have to set the format
for the output because ecasound defaults to 16,2,44100. You don't need
to set the format for the inputs if they are .wavs because ecasound will
read the info from the wav headers. Then for chain 2 you route channel
one to channel two (-erc:1,2), and mute channel 1 (-eac:0,1).
-ERic Rz.
4:38 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 2004-09-09 at 11:02, Eric Dantan Rzewnicki wrote:
ecasound!
ecasound -a:1 -i:mono1.wav \
-a:2 -i:mono2.wav \
-a:1,2 -f:24,2,96000 -o:stereo.wav \
-a:2 -erc:1,2 -eac:0,1
Do that all on one line. The returns are escaped to make it easier to
read. This creates 2 chains, one for each input mono wav file. Both
chains are connected to the same output file. You have to set the format
for the output because ecasound defaults to 16,2,44100. You don't need
to set the format for the inputs if they are .wavs because ecasound will
read the info from the wav headers. Then for chain 2 you route channel
one to channel two (-erc:1,2), and mute channel 1 (-eac:0,1).
-ERic Rz.
Thanks. I'll give that a try. At least that handles step one, assuming
no audio problems are created.
cheers,
Mark
5:49 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, Sep 09, 2004 at 11:21:35AM -0700, Mark Knecht wrote:
On Thu, 2004-09-09 at 11:02, Eric Dantan Rzewnicki
wrote:
If step 2 was downsampling to 16 bit 44.1kHz, then just change the -f
above to: -f:16,2,44100 ... sorry I missed that bit.
The internal sample rate conversion is linear interpolation, I think,
which works ok. But, if you have libsamplerate installed ecasound can
use that instead for higher quality conversion. libsamplerate support is
enabled at ./configure time. I always install ecasound from source, so
I'm not sure how the various distros compile and package it. I suppose
on one of your gentoo boxes that should be no problem.
-Eric Rz.
ecasound!
ecasound -a:1 -i:mono1.wav \
-a:2 -i:mono2.wav \
-a:1,2 -f:24,2,96000 -o:stereo.wav \
-a:2 -erc:1,2 -eac:0,1
Do that all on one line. The returns are escaped to make it easier to
read. This creates 2 chains, one for each input mono wav file. Both
chains are connected to the same output file. You have to set the format
for the output because ecasound defaults to 16,2,44100. You don't need
to set the format for the inputs if they are .wavs because ecasound will
read the info from the wav headers. Then for chain 2 you route channel
one to channel two (-erc:1,2), and mute channel 1 (-eac:0,1).
-ERic Rz.
Thanks. I'll give that a try. At least that handles step one,
assuming
no audio problems are created.
6:39 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 2004-09-09 at 12:32, Eric Dantan Rzewnicki wrote:
On Thu, Sep 09, 2004 at 11:21:35AM -0700, Mark Knecht
wrote:
Eric,
I do have libsamplerate installed on my laptop here in the office.
Since all of this is just file manipulation I can do everything here and
then listen elsewhere. (Onboard sound works, but is not so good.
Interesting bass distortion problems even at VERY low levels when
playing with Jamin this morning. I hope that's not Jamin itself...)
How would I be sure that my emerge of ecasound uses libsamplerate
capabilities? Can this be observed at runtime? Or in the output files
somehow? Anywhere else?
Anyway, If I'm going to downsize from 24-bits to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Thanks all,
Mark
On Thu, 2004-09-09 at 11:02, Eric Dantan
Rzewnicki wrote:
If step 2 was downsampling to 16 bit 44.1kHz, then just change the -f
above to: -f:16,2,44100 ... sorry I missed that bit.
The internal sample rate conversion is linear interpolation, I think,
which works ok. But, if you have libsamplerate installed ecasound can
use that instead for higher quality conversion. libsamplerate support is
enabled at ./configure time. I always install ecasound from source, so
I'm not sure how the various distros compile and package it. I suppose
on one of your gentoo boxes that should be no problem.
-Eric Rz. ecasound!
ecasound -a:1 -i:mono1.wav \
-a:2 -i:mono2.wav \
-a:1,2 -f:24,2,96000 -o:stereo.wav \
-a:2 -erc:1,2 -eac:0,1
Do that all on one line. The returns are escaped to make it easier to
read. This creates 2 chains, one for each input mono wav file. Both
chains are connected to the same output file. You have to set the format
for the output because ecasound defaults to 16,2,44100. You don't need
to set the format for the inputs if they are .wavs because ecasound will
read the info from the wav headers. Then for chain 2 you route channel
one to channel two (-erc:1,2), and mute channel 1 (-eac:0,1).
-ERic Rz.
Thanks. I'll give that a try. At least that handles step one,
assuming
no audio problems are created. 
6:51 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, Sep 09, 2004 at 01:22:45 -0700, Mark Knecht wrote:
I do have libsamplerate installed on my laptop here
in the office.
Since all of this is just file manipulation I can do everything here and
then listen elsewhere. (Onboard sound works, but is not so good.
Interesting bass distortion problems even at VERY low levels when
playing with Jamin this morning. I hope that's not Jamin itself...)
JAMin has had problem with bass distortion in the past, if it is JAMin
please supply some test material and I'l look into it - OTOH if youre
using linear interpolation - stop it! :)
Anyway, If I'm going to downsize from 24-bits
to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Thats right. Soundcard only.
- Steve
7:13 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 2004-09-09 at 13:35, Steve Harris wrote:
So is there any solution to dithering a final output? I think we talked
about this recently, but the answer eludes me. If there is a dither
plugin I couldn't find it. I was supposing that the 24-bit output could
be dithered to 16 bits but saved as 24-bits, and then maybe the bottom 8
bits just get stripped at the command line or something like that.
I don't hear the word dither come up as much around here as other
places...
thanks for your help.
On Thu, Sep 09, 2004 at 01:22:45 -0700, Mark Knecht
wrote:
Steve,
I'll fire it back up this afternoon and look for linear interpolation
options. I was simply in a default set of settings, other than using the
solo button to hear the individual bands. The bass bands had very little
energy, according to the spectrum analyzer, and the overall levels were
set VERY low - like -20db, but I was still hearing noticible breakup.
Maybe I can just record it in Ardour and send you that?
One other possibility, I suppose, is that the problem is there on the
original tracks. I did the tracking at 96KHz in Pro Tools and hadn't
noticed any problems, but I wasn't searching either. Import to Ardour
converts them to 44.1KHz as my laptop doesn't do 96K. Possibly the
problem is in the conversion, or in the originals but masked somehow. To
check that I'll have to listen closely in Pro Tools this evening.
I do have libsamplerate installed on my laptop
here in the office.
Since all of this is just file manipulation I can do everything here and
then listen elsewhere. (Onboard sound works, but is not so good.
Interesting bass distortion problems even at VERY low levels when
playing with Jamin this morning. I hope that's not Jamin itself...)
JAMin has had problem with bass distortion in the past, if it is JAMin
please supply some test material and I'l look into it - OTOH if youre
using linear interpolation - stop it! :) Anyway, If I'm going to downsize from
24-bits to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Thats right. Soundcard only.
- Steve
10 Sep
10 Sep
7:02 a.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, Sep 09, 2004 at 01:57:17 -0700, Mark Knecht wrote:
So is there any solution to dithering a final output? I think we talked
about this recently, but the answer eludes me. If there is a dither
plugin I couldn't find it. I was supposing that the 24-bit output could
be dithered to 16 bits but saved as 24-bits, and then maybe the bottom 8
bits just get stripped at the command line or something like that.
Ardour can dither, so if you record the output of JAMin into ardour and
export from there you should be able to save down as 16bits.
- Steve
On Thu, 2004-09-09 at 13:35, Steve Harris wrote:
Oh, I didn't mean in JAMin, JAMin doesnt do interpolation, I meant in
opther parts of the chain.
On Thu, Sep 09, 2004 at 01:22:45 -0700, Mark
Knecht wrote:
Steve,
I'll fire it back up this afternoon and look for linear interpolation
options. I was simply in a default set of settings, other than using the
solo button to hear the individual bands. The bass bands had very little
energy, according to the spectrum analyzer, and the overall levels were
set VERY low - like -20db, but I was still hearing noticible breakup. I do have libsamplerate installed on my laptop
here in the office.
Since all of this is just file manipulation I can do everything here and
then listen elsewhere. (Onboard sound works, but is not so good.
Interesting bass distortion problems even at VERY low levels when
playing with Jamin this morning. I hope that's not Jamin itself...)
JAMin has had problem with bass distortion in the past, if it is JAMin
please supply some test material and I'l look into it - OTOH if youre
using linear interpolation - stop it! :) Maybe I can just record it in Ardour and send you
that?
The actual sound wont help, I really need the ingredients to recreate it
I'm afraid.
One other possibility, I suppose, is that the
problem is there on the
original tracks. I did the tracking at 96KHz in Pro Tools and hadn't
noticed any problems, but I wasn't searching either. Import to Ardour
converts them to 44.1KHz as my laptop doesn't do 96K. Possibly the
problem is in the conversion, or in the originals but masked somehow. To
check that I'll have to listen closely in Pro Tools this evening.
Sometimes bringing up the average level in JAMin can be just enough to
reveal problem that were masked by other frequencies before, but dont rul
out a JAMin bug.
Anyway, If I'm going to downsize from 24-bits
to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Thats right. Soundcard only.
6:43 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, 2004-09-10 at 01:46, Steve Harris wrote:
Thanks for the pointer Steve. I have managed to record some track in
Ardour through Jamin and have exported them with dithering. I'll have to
wait to do any critical listening but at least the process flow is
working.
Much appreciated!
Ale or stout, it's on me.
Cheers,
Mark
So is there
any solution to dithering a final output? I think we talked
about this recently, but the answer eludes me. If there is a dither
plugin I couldn't find it. I was supposing that the 24-bit output could
be dithered to 16 bits but saved as 24-bits, and then maybe the bottom 8
bits just get stripped at the command line or something like that.
Ardour can dither, so if you record the output of JAMin into ardour and
export from there you should be able to save down as 16bits.
- Steve
9 Sep
9 Sep
7:24 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, Sep 09, 2004 at 01:22:45PM -0700, Mark Knecht wrote:
On Thu, 2004-09-09 at 12:32, Eric Dantan Rzewnicki
wrote:
Here is one way to be sure ecasound uses libsamplerate:
ecasound -f:16,2,44100 -i:resample-hq,auto,hi-quality.wav -o:lower-quality.wav
The resample-hq input object type, with the auto flag, is a special type
that causes ecasound to automatically read the format of the input .wav
and resample it using libsamplerate to match the format set for the rest
of the chain. I imagine the above command will fail if ecasound is not
compiled with src support. There is a similar resample object that uses
the internal linear interpolation. For both object types auto can be
replaced with the actual sample format if your input file format lacks
that info. One example would be raw input files.
Rewriting the earlier example, your whole conversion can be done with
one commandline:
ecasound -f:16,2,44100 \
-a:1 -i:resample-hq,auto,mono1.wav \
-a:2 -i:resample-hq,auto,mono2.wav \
-a:1,2 -o:stereo.wav \
-a:2 -erc:1,2 -eac:0,1
(untested, but as far as I know, it should work)
-Eric Rz.
On Thu, Sep 09, 2004 at 11:21:35AM -0700, Mark
Knecht wrote:
Eric,
How would I be sure that my emerge of ecasound uses libsamplerate
capabilities? Can this be observed at runtime? Or in the output files
somehow? Anywhere else?
Anyway, If I'm going to downsize from 24-bits to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone? On Thu, 2004-09-09 at 11:02, Eric Dantan
Rzewnicki wrote:
If step 2 was downsampling to 16 bit 44.1kHz, then
just change the -f
above to: -f:16,2,44100 ... sorry I missed that bit. ecasound!
ecasound -a:1 -i:mono1.wav \
-a:2 -i:mono2.wav \
-a:1,2 -f:24,2,96000 -o:stereo.wav \
-a:2 -erc:1,2 -eac:0,1
Thanks. I'll give that a try. At least that handles
step one, assuming
no audio problems are created.
7:50 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 2004-09-09 at 14:07, Eric Dantan Rzewnicki wrote:
Here is one way to be sure ecasound uses
libsamplerate:
ecasound -f:16,2,44100 -i:resample-hq,auto,hi-quality.wav -o:lower-quality.wav
The resample-hq input object type, with the auto flag, is a special type
that causes ecasound to automatically read the format of the input .wav
and resample it using libsamplerate to match the format set for the rest
of the chain. I imagine the above command will fail if ecasound is not
compiled with src support. There is a similar resample object that uses
the internal linear interpolation. For both object types auto can be
replaced with the actual sample format if your input file format lacks
that info. One example would be raw input files.
It ran, but it did not apparently downsample. Granted I took a mono file
as input and wrote a stereo file out, but the output sample rate is
still reported as 96K. I will try your 2-mono to stereo example in a
minute.
ecasound -f:16,2,44100 -i:resample-hq,auto,
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav -o:TEST1.wav
****************************************************************************
* ecasound v2.3.3 (C) 1997-2004 Kai
Vehmanen
****************************************************************************
- [ Session created ]
------------------------------------------------------
- [ Chainsetup created (cmdline) ]
-----------------------------------------
(eca-chainsetup-parser) Note! Interpreting option
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav as
-i:data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav.
- [ Connecting chainsetup ]
------------------------------------------------
(eca-chainsetup) 'nonrt' buffering mode selected.
Warning: DBC_CHECK failed - "samples_per_second() > 0", audioio.cpp,
195.
Warning: DBC_CHECK failed - "samples_per_second() > 0", audioio.cpp,
195.
(eca-chainsetup) Audio object "", mode "read".
(audio-io) Format: s16_le, channels 2, srate 44100, interleaved.
(eca-chainsetup) Audio object
"data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav", mode "read".
(audio-io) Format: s24_le, channels 1, srate 96000, interleaved.
(eca-chainsetup) Audio object "TEST1.wav", mode "read/write".
(audio-io) Format: s16_le, channels 2, srate 96000, interleaved.
- [ Chainsetup connected ]
-------------------------------------------------
(eca-controller) Connected chainsetup: "command-line-setup".
- [ Controller/Starting batch processing ]
---------------------------------
- [ Engine init - Driver start ]
-------------------------------------------
- [ Controller/Batch processing finished ]
---------------------------------
- [ Engine exiting ]
-------------------------------------------------------
(eca-controller) Disconnecting chainsetup: "command-line-setup".
- [ Chainsetup disconnected ]
----------------------------------------------
flash mark $
flash mark $ sndfile-info TEST1.wav
Version : libsndfile-1.0.10
========================================
File : TEST1.wav
Length : 81780116
RIFF : 81780104
WAVE
fmt : 16
Format : 0x1 => WAVE_FORMAT_PCM
Channels : 2
Sample Rate : 96000
Block Align : 4
Bit Width : 16
Bytes/sec : 384000
data : 81780072
End
----------------------------------------
Sample Rate : 96000
Frames : 20445018
Channels : 2
Format : 0x00010002
Sections : 1
Seekable : TRUE
Duration : 00:03:32.968
Signal Max : 30318 (-0.67 dB)
flash mark $
flash mark $ sndfile-info
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav
Version : libsndfile-1.0.10
========================================
File : data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav
Length : 61336078
RIFF : 61336070
WAVE
bext : 602
fmt : 16
Format : 0x1 => WAVE_FORMAT_PCM
Channels : 1
Sample Rate : 96000
Block Align : 3
Bit Width : 24
Bytes/sec : 288000
*** minf : 16 (unknown marker)
*** elmo : 338 (unknown marker)
data : 61175574
*** regn : 92 (unknown marker)
*** ovwf : 159340 (unknown marker)
*** umid : 24 (unknown marker)
End
----------------------------------------
Sample Rate : 96000
Frames : 20391858
Channels : 1
Format : 0x00010003
Sections : 1
Seekable : TRUE
Duration : 00:03:32.415
Signal Max : 4.20391e+06 (-6.00 dB)
flash mark $
8:44 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 9 Sep 2004, Mark Knecht wrote:
ecasound -f:16,2,44100 -i:resample-hq,auto,
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav -o:TEST1.wav
Did TEST1.wav already exist before you ran the above? If it did, the
format -f:16,2,44100 will not be used. Otherwise it should work normally
(just tested with the same version of ecasound)...
(eca-chainsetup) Audio object "", mode
"read".
(audio-io) Format: s16_le, channels 2, srate 44100, interleaved.
(eca-chainsetup) Audio object
"data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav", mode "read".
(audio-io) Format: s24_le, channels 1, srate 96000, interleaved.
(eca-chainsetup) Audio object "TEST1.wav", mode "read/write".
(audio-io) Format: s16_le, channels 2, srate 96000, interleaved.
Hmm, it shouldn't show three audio objects here. Make sure you don't have
any whitespace in "-i:resample-hw,auto,foobar.wav".
--
http://www.eca.cx
Audio software for Linux!
9:03 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, Sep 10, 2004 at 01:27:38AM +0300, Kai Vehmanen wrote:
On Thu, 9 Sep 2004, Mark Knecht wrote:
oops. I was too slow.
ecasound -f:16,2,44100 -i:resample-hq,auto,
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav -o:TEST1.wav
Did TEST1.wav already
exist before you ran the above? If it did, the
format -f:16,2,44100 will not be used. Otherwise it should work normally
(just tested with the same version of ecasound)...
(eca-chainsetup) Audio object "", mode
"read".
(audio-io) Format: s16_le, channels 2, srate 44100, interleaved.
(eca-chainsetup) Audio object
"data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav", mode "read".
(audio-io) Format: s24_le, channels 1, srate 96000, interleaved.
(eca-chainsetup) Audio object "TEST1.wav", mode "read/write".
(audio-io) Format: s16_le, channels 2, srate 96000, interleaved.
Hmm, it
shouldn't show three audio objects here. Make sure you don't have
any whitespace in "-i:resample-hw,auto,foobar.wav".
9:01 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, Sep 09, 2004 at 02:33:57PM -0700, Mark Knecht wrote:
On Thu, 2004-09-09 at 14:07, Eric Dantan Rzewnicki
wrote:
I think you just have an extra line break or space in your commandline.
-Eric Rz.
Here is one way to be sure ecasound uses
libsamplerate:
ecasound -f:16,2,44100 -i:resample-hq,auto,hi-quality.wav -o:lower-quality.wav
The resample-hq input object type, with the auto flag, is a special type
that causes ecasound to automatically read the format of the input .wav
and resample it using libsamplerate to match the format set for the rest
of the chain. I imagine the above command will fail if ecasound is not
compiled with src support. There is a similar resample object that uses
the internal linear interpolation. For both object types auto can be
replaced with the actual sample format if your input file format lacks
that info. One example would be raw input files.
It ran, but it did not apparently
downsample. Granted I took a mono file
as input and wrote a stereo file out, but the output sample rate is
still reported as 96K. I will try your 2-mono to stereo example in a
minute.
ecasound -f:16,2,44100 -i:resample-hq,auto,
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav -o:TEST1.wav
****************************************************************************
* ecasound v2.3.3 (C) 1997-2004 Kai
Vehmanen
****************************************************************************
- [ Session created ]
------------------------------------------------------
- [ Chainsetup created (cmdline) ]
-----------------------------------------
(eca-chainsetup-parser) Note! Interpreting option
^^^^^^^^^^^^^^^^^^^^^^^^^
data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav as
-i:data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav.
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
This doesn't look correct. The parser saw the "data/.../...wav"
separately from the "-i:resample-hq,auto,". Did you perhaps have a space
in there by accident? (or maybe in my example?)
This message typically comes out if you don't specify a -i: option at
all. The commandline parser defaults to interpreting the first thing
that looks like a filename as the input.
- [ Connecting chainsetup ]
------------------------------------------------
(eca-chainsetup) 'nonrt' buffering mode selected.
Warning: DBC_CHECK failed - "samples_per_second() > 0", audioio.cpp,
195.
Warning: DBC_CHECK failed - "samples_per_second() > 0", audioio.cpp,
195.
^^^^^^^
These point to it trying to autodetect the sample rate and not finding
anything valid, I think.
(eca-chainsetup) Audio object "", mode
"read".
^^^
This looks like it interpreted your -i:resample-hq,auto, as having an
empty string for the filename ...
(audio-io) Format: s16_le, channels 2, srate 44100,
interleaved.
and since it couldn't find a samplerate in the non-existent file it
went
with the sample rate you set with the -f:16,2,44100 at the start of your
commandline. Which doesn't matter, because this file doesn't exist
anyway.
(eca-chainsetup) Audio object
"data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav", mode "read".
(audio-io) Format: s24_le, channels 1, srate 96000, interleaved.
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
and then here it saw the filename by itself as a second input and read
the format from the wav headers. Since the resample-hq type didn't stick
to this file, the sample rate for the rest of the chain is overridden by
that in the wav file ...
(eca-chainsetup) Audio object "TEST1.wav",
mode "read/write".
(audio-io) Format: s16_le, channels 2, srate 96000, interleaved.
- [ Chainsetup connected ]
leading to your output file inheritting that sample
rate, and your
operation basically doing nothing for you other than down sample to 16
bits.
Strictly speaking, I don't think the resample-hq or resample object type
is necessary to get _some_ sort of resampling. It's an explicit way to
ensure the type of resampling you want is used. Or something along those
lines. Hopefully Kai will correct me if I'm mixed up, but I think I'm at
least very close.
8:29 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 9 Sep 2004, Mark Knecht wrote:
How would I be sure that my emerge of ecasound uses
libsamplerate
capabilities? Can this be observed at runtime? Or in the output files
somehow? Anywhere else?
ldd `which ecasound` |grep libsamplerate
Anyway, If I'm going to downsize from 24-bits
to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Unfortunately no. Adding dithering has been on the todo-list for ages,
but still not implemented. :(
Also, Ecasound's 24bit is really 24bit, not the 24-in-32bit. This is
usually not a problem, just as long you are aware of it.
--
http://www.eca.cx
Audio software for Linux!
9:20 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Thu, 2004-09-09 at 15:13, Kai Vehmanen wrote:
I can understand having things like this on your todo list. I guess I'm
now wondering if anyone running Linux is dithering their files before
they make CDs? Possibly no one is doing this? I'm just very used to
doing it on the Pro Tools side, and I still can, but it would be nice to
set up a mastering environment under Alsa and just stay in Linux all the
way to burning the CD if possible. I'd hate to go back to Windows just
for the dithering step. (I might have to if Linux dithering isn't up to
the task, as well it might not be if no one is really using it. We'll
see. (Or hear...) ;-)
Anyway, it's a few days away and not a huge amount of work if I have to.
On Thu, 9 Sep 2004, Mark Knecht wrote:
flash mark $ ldd `which ecasound` |grep libsamplerate
libsamplerate.so.0 => /usr/lib/libsamplerate.so.0 (0x40067000)
flash mark $
Oops - I did have white space in the command line:
<SNIP>
-i:resample-hq,auto, data/GTSB-96K...
<SNIP>
flash mark $ ecasound -f:16,2,44100
-i:resample-hq,auto,data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav
-o:TEST1.wav
****************************************************************************
* ecasound v2.3.3 (C) 1997-2004 Kai
Vehmanen
****************************************************************************
- [ Session created ]
------------------------------------------------------
- [ Chainsetup created (cmdline) ]
-----------------------------------------
- [ Connecting chainsetup ]
------------------------------------------------
(eca-chainsetup) 'nonrt' buffering mode selected.
(eca-chainsetup) Audio object
"data/GTSB-96K-24bit-Audio/T01-Startdust.L.wav", mode "read".
(audio-io) Format: s24_le, channels 1, srate 44100, interleaved.
(eca-chainsetup) Audio object "TEST1.wav", mode "read/write".
(audio-io) Format: s16_le, channels 2, srate 44100, interleaved.
- [ Chainsetup connected ]
-------------------------------------------------
(eca-controller) Connected chainsetup: "command-line-setup".
- [ Controller/Starting batch processing ]
---------------------------------
- [ Engine init - Driver start ]
-------------------------------------------
Warning: DBC_CHECK failed - "impl_repp->rt_lock_rep != true",
samplebuffer.cpp, 850.
- [ Controller/Batch processing finished ]
---------------------------------
- [ Engine exiting ]
-------------------------------------------------------
(eca-controller) Disconnecting chainsetup: "command-line-setup".
- [ Chainsetup disconnected ]
----------------------------------------------
flash mark $
and the output is corretly 44.1K
flash mark $ sndfile-info TEST1.wav
Version : libsndfile-1.0.10
========================================
File : TEST1.wav
Length : 37533024
RIFF : 37533012
WAVE
fmt : 16
Format : 0x1 => WAVE_FORMAT_PCM
Channels : 2
Sample Rate : 44100
Block Align : 4
Bit Width : 16
Bytes/sec : 176400
data : 37532980
End
----------------------------------------
Sample Rate : 44100
Frames : 9383245
Channels : 2
Format : 0x00010002
Sections : 1
Seekable : TRUE
Duration : 00:03:32.771
Signal Max : 27180 (-1.62 dB)
flash mark $
How would I be sure that my emerge of ecasound
uses libsamplerate
capabilities? Can this be observed at runtime? Or in the output files
somehow? Anywhere else?
ldd `which ecasound` |grep libsamplerate Anyway, If I'm going to downsize from
24-bits to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Unfortunately no. Adding dithering has been on the todo-list for ages,
but still not implemented. :(
Also, Ecasound's 24bit is really 24bit, not the 24-in-32bit. This is
usually not a problem, just as long you are aware of it.
I don't think that's a problem for me as long as I can send it back to
Pro Tools for dithering. I'll test that tonight or tomorrow. Possibly
I'll have to run sndfile-convert on the output before I go back? We'll
see...
Thanks for your help!
- Mark
10 Sep
10 Sep
7:05 a.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, Sep 10, 2004 at 01:13:16 +0300, Kai Vehmanen wrote:
On Thu, 9 Sep 2004, Mark Knecht wrote:
I'm in the process of making a (trivial) libgdither that has a simple API
to it. Currently it works but I want to add a couple of API features. Its
based off the code in ardour, but much faster and there some bugfixes.
- Steve
How would I be sure that my emerge of ecasound
uses libsamplerate
capabilities? Can this be observed at runtime? Or in the output files
somehow? Anywhere else?
ldd `which ecasound` |grep libsamplerate
Anyway, If I'm going to downsize from
24-bits to 16-bits then I want
to add dithering, but I see no mention of dithering in man ecasound. I
know jack has an option to dither it's outputs, but from some dark
remote part of my memory I seem to remember that this only effects the
output going to the sound card and not what's going to a file. Comments
anyone?
Unfortunately no. Adding dithering has been on the todo-list for ages,
but still not implemented. :( 
7:33 a.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, 10 Sep 2004 09:48:35 +0100
Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> wrote:
I'm in the process of making a (trivial)
libgdither that has a simple API
to it. Currently it works but I want to add a couple of API features. Its
based off the code in ardour, but much faster and there some bugfixes.
Cool. If you release it under the LGPL, I'll probably snaffle the
code and add it to libsndfile WIFT (When I Find Time) :-).
Erik
--
+-----------------------------------------------------------+
Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
GPLG
GPLGPLGP
GPLGPLGPLGP
GPLGP
GPL MICROSOFT
GPLGP
GPLGPLGPLGP
GPLGPLGPL
GPLGPL
9:27 a.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, Sep 10, 2004 at 07:17:06 +1000, Erik de Castro Lopo wrote:
On Fri, 10 Sep 2004 09:48:35 +0100
Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> wrote:
Its currently GPL, but just because thats my default - if it causes any
Free Software project problems I'm happy to dual licence it, or whatever.
- Steve
I'm in the process of making a (trivial)
libgdither that has a simple API
to it. Currently it works but I want to add a couple of API features. Its
based off the code in ardour, but much faster and there some bugfixes.
Cool. If you release it under the LGPL, I'll probably snaffle the
code and add it to libsndfile WIFT (When I Find Time) :-).
5:02 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, Sep 10, 2004 at 07:17:06PM +1000, Erik de Castro Lopo wrote:
On Fri, 10 Sep 2004 09:48:35 +0100
Steve Harris <S.W.Harris(a)ecs.soton.ac.uk> wrote:
Does libsamplerate dither at all? or is it only concerned with sample
rate and doesn't care about sample width?
-Eric Rz.
I'm in the process of making a (trivial)
libgdither that has a simple API
to it. Currently it works but I want to add a couple of API features. Its
based off the code in ardour, but much faster and there some bugfixes.
Cool. If
you release it under the LGPL, I'll probably snaffle the
code and add it to libsndfile WIFT (When I Find Time) :-).
8:11 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, 10 Sep 2004 14:46:11 -0400
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> wrote:
Does libsamplerate dither at all? or is it only
concerned with sample
rate and doesn't care about sample width?
libsamplerate only works on float data. Float data in and float
data out. It doesn't care about sample width.
Erik
--
+-----------------------------------------------------------+
Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
Fundamentalist : Someone who is colour blind and yet wants everyone
else to see the world with the same lack of colour.
9:07 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Sat, Sep 11, 2004 at 07:54:47AM +1000, Erik de Castro Lopo wrote:
On Fri, 10 Sep 2004 14:46:11 -0400
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> wrote:
Ok. thanks.
You always have great quotes in your sig. Do you harvest these manually
and put them in on some random rotation?
-Eric Rz.
Does libsamplerate dither at all? or is it only
concerned with sample
rate and doesn't care about sample width?
libsamplerate only works on float
data. Float data in and float
data out. It doesn't care about sample width.
9:41 p.m.
New subject: [linux-audio-user] Wave file conversion (multi-mono->stereo, then downsample/downsize, then dither)
On Fri, 10 Sep 2004 18:50:47 -0400
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> wrote:
On Sat, Sep 11, 2004 at 07:54:47AM +1000, Erik de
Castro Lopo wrote:
Thanks.
On Fri, 10 Sep 2004 14:46:11 -0400
Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> wrote:
Ok. thanks.
You always have great quotes in your sig. Does libsamplerate dither at all? or is it only
concerned with sample
rate and doesn't care about sample width?
libsamplerate only works on float
data. Float data in and float
data out. It doesn't care about sample width. Do you harvest these manually
Yep.
and put them in on some random rotation?
and yep.
Erik
--
+-----------------------------------------------------------+
Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
"When your hammer is C++, everything begins to look like a thumb."
-- Steve Haflich, in comp.lang.c++
--
+-----------------------------------------------------------+
Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
"C is a programming language. C++ is a cult."
-- David Parsons in comp.os.linux.development.apps
7323
days inactive
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days old
linux-audio-user@lists.linuxaudio.org
22 comments
6 participants
participants (6)
-
Eric Dantan Rzewnicki -
Erik de Castro Lopo -
Kai Vehmanen -
Mark Knecht -
Mark Knecht -
Steve Harris