Displaying 20 results from an estimated 3000 matches similar to: "Encoding audio sampled at 44.1 khz?"
2007 Mar 22
1
[SPAM] RE: Encoding audio sampled at 44.1 khz?
________________________________
Hi David,
Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex.
But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below).
The test data I had was a file sampled
2007 Mar 22
0
Encoding audio sampled at 44.1 khz?
Hi Peter,
Have you considered resampling the raw 44.1kHz stereo source files using
a program such as http://audacity.sourceforge.net/, say to 16kHz mono or
32kHz mono, and then using the wideband or ultrawideband speex modes to
encode the result?
Alternatively, if you want to programmatically do the resampling
yourself, you could try the new resampling API in the svn head of speex,
or the GPL
2007 Apr 02
1
Problems with stereo data
Hi all,
I have a problem when I am encoding (or decoding) stereo audio.
With mono data, things are fine and everything works without any problems.
When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong?
Below is the code, first, of the encoder and next that of the decoder. Any information or
2009 Mar 17
2
Resample UltraWideBand to NarrowBand
Hi List,
Now I will send to you more specific what I am trying to do.
I have one Asterisk Channel where receives Midia Frames in the codecs
format: Speex UltraWideBand and Speex NarrowBand.
When I use Speex NarrowBand the Asterisk is able to convert this frame to
G711.
But when I use Speex UltraWideBand the Asterisk don't convert it.
Then I need in my Asterisk Channel Source include the Speex
2023 Feb 22
1
Change 48 khz sample rate limit
You asked in the Vorbis list, but your text only mentions OGG. The
codec commonly used in OGG containers that is limited to 48 khz is
Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead
of Vorbis)?
Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis
file with the following command:
$ oggenc --resample 192000 input.wav
Of course, if your original material is
2011 Nov 10
2
Removing numbers from a list
I am using gsub to remove numbers for each element of a list. Code is given
below.
testList <- list("this contains a number 1000","this does not contain")
removeNumbers <- function(X)
{
gsub("\\d","",X)
}
outputList <- lapply(testList,removeNumbers)
However, when I try to find the number of words in outputList as follows
2004 Apr 02
2
resampling to 48 kHz
One thing that has always bothered me about the ogg
format is the distortion of high frequency sounds -
even at data rates as high 128 and 160 kbps. I find
the best way around this is to resample the wav file
to 48 kHz (using SoundForge 6.0) before encoding
(using CDex) to ogg. It takes a while, and adds a lot
of extra wear and tear on my drive, but what a
difference! The result is an 80k ogg file
2023 Feb 22
2
Change 48 khz sample rate limit
Hi!, I wondering if It's possible to change 48khz sample rate limit?,
I'm Planing to encode with OGG codec a audio signal but I need that
OGG Encoder works with 192khz of sample rate.
It's Possible?
Any Suggestions?
2014 Aug 14
1
Encoder example for 24-bit files
On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.mail at gmail.com> wrote:
> Jose Pablo Carballo <jose.carballo at ridgerun.com> wrote:
>
>> - channels = 2;
>> - bps = 16;
>> + channels = ((unsigned)buffer[23] << 8) | buffer[22];
>> + bps = ((unsigned)buffer[35] << 8) | buffer[34];
>> total_samples = (((((((unsigned)buffer[43] << 8)
2006 Dec 02
2
encoding wave files into ogg vorbis?
Hi everyone,
I am a newbe with the Ogg Vorbis libraries.
I am trying to create a program in C++ that will allow me to take a 16
bit, stereo wave file at 44.1 khz and encode it into ogg vorbis. I
have looked at the encoder example in the vorbis SDK, but I did not
quite understand it.
My question, therefore, is how to encode such a stream into vorbis? A
simple tutorial would be very much
2014 Aug 14
6
Encoder example for 24-bit files
Hi,
In the last days I've been taking as reference the example found in
examples/c/encode/file/main.c. With it I've been able to encode a 2ch,
16 bps, 44100 sample rate input WAV file to a FLAC file.
Now I've been trying to modify this example to encode a 2ch, 24 bps,
96000 sample rate WAV file. I have to say I'm a bit lost on how I
should read the input file in this case, and
2007 Sep 30
9
Problems with testing nested routes using mocking
Hello forum
I have there to files
#----- virtual_host_controller.rb
class VirtualHostsController < ApplicationController
before_filter :capture_domain
# GET /domain/1/virtual_hosts/1
def show
@virtual_host = @domain.virtual_hosts.find(params[:id])
respond_to do |format|
format.html # show.rhtml
end
end
private
def capture_domain
if
2006 May 29
7
re-coding a sizable PHP app in rails
Hello,
I am the developer of a fairly major PHP app. It has the full
compliment of web application goodies -- email, batch processes, cc
transactions, multi-level authentication, security, content management,
curl-type interaction with other applications, etc. I am obsessed with
the idea of re-doing a year and a half of work in Rails and the more I
learn the worse my itch gets. I feel
2016 Mar 15
3
Question on opus_decoder output sampling rate
Hi, another question on the same topic
Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even
more than the Opus encoder)
While Speex at 48kHz is just fine.
I wonder any alternate solutions or ideas ?
Improve it, look for alternate solution ...
I am guessing the NEON optimization are still used for both, etc.
On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at
2007 Dec 02
3
Setting SSH timeout
i'm trying to disconnect idle users from my system by editing
/etc/ssh/sshd_config
i have set
TCPKeepAlive no
ClientAliveInterval 2
and restarting sshd services /etc/rc.d/sshd restart
but it still wont disconnect any idle client
any advice is highly appreciated
areadamai
freebsd user
2008 Nov 14
3
SPEEX on iPhone ?
----- Original Message -----
From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com>
To: "Vincent Burel" <vincent.burel at vb-audio.com>
Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc
Valin" <jean-marc.valin at usherbrooke.ca>
Sent: Thursday, November 13, 2008 11:31 PM
Subject: Re:
2008 May 28
2
FFT Resampler
Attached is a snapshot of work-in-progress of a FFT based resampler. At
the moment it works in floating point only, and only basic quality
inspection has been done.
Some benchmarks comparing the filter-based resampler at Q3 with the FFT
resampler with overlap = in_len / 2, using 20ms chunks of data. (-O3
-ffast-math, FFTW3, gcc 4.3.0 on x86_64)
16=>48: 59us vs 19us
16=>44.1: 204us vs
2017 Aug 19
2
bootstrap subject resampling: resampled subject codes surface as list/vector indices
I'm implementing a custom bootstrap resampling procedure in R. This
procedure resamples clusters of data points obtained by different
subjects in an experiment. Since the bootstrap samples need to have the
same size as the original dataset, `target.set.size`, I select speakers
compute their data point contributions to make sure I have a set of the
right size.
set.seed(1)
2004 Aug 06
7
question on downsampling
Hi,
Maybe a bit off topic for this list, bt anyway.
I have received several feature requests for DarkIce to support
downsampling of the audio input before passing it to lame or ogg vorbis.
For example the audio read from the soundcard would be 44.1kHz, and lame
would get it at 22.05kHz.
I figure two ways of doing this:
1. For lame, one can specify the input and the desired mp3 sampling
rate,
2002 Dec 22
1
oggdropXPd.exe crashes when resampling is activated
I need to encode a WAV to OGG but must resample to 22050 samples per second,
but can't do that with oggdropXPd.exe. The application always crashes.
What I do is:
Start oggdropXPd.exe. Right-click on fish, choosing 'Encoding Options'.
Under Advanced Options, I set Re-sample to 22050 samples per second. Click
'Accept'. Drag WAV on fish. In the directory where the WAV originates,