similar to: 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

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

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

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

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

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

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

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?

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)

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

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

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

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

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

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

----- 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:

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

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)

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,

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,