similar to: FLAC From 32 Bits

Hello. I create FLAC file decoding, processing and playing program and have the following question : how to convert FLAC 16 bit file data to 32 bit float buffer for CPU processing? I've already inplemented sound playing and tested it with sine wave - it works without problems; I even made writing into decoding buffer values of sine wave, instead of decoded FLAC file data, and it also works

The following block in os.h needs to be changed for vorbis to work under Borland: Original: #ifndef __GNUC__ #ifdef _WIN32 # define alloca(x) (_alloca(x)) # define rint(x) (floor((x)+0.5)) #endif #endif New: #ifndef __GNUC__ #ifdef _WIN32 # ifdef __BORLANDC__ # include <malloc.h> # else # define alloca(x) (_alloca(x)) # endif # define rint(x) (floor((x)+0.5)) #endif

The following block of code produces an exception. The testnew.ogg file is 10 seconds, containing pure silence encoded using mode E: http://www.cs.mun.ca/~chris3/testnew.ogg Other files and modes can cause the exception, but many files work fine. #include <stdio.h> #include "vorbis/vorbisfile.h" int main() { OggVorbis_File ov, ov2; FILE *f, *f2; f

Hello, I am author of a Morse code training software written in Perl at http://starling.ws/morse Currently I output to *.wav and, if on Unix, convert to *.ogg. And that's okay for a trainer. But I have desire to upgrade the project to a real-time communication tool...or at least a Morse code audio mail routine. And I should like to avoid *.wav. I'd like some means of writing a pure

My tuneR sawtooth wave function generator is broken. When I use the sine function, I get exactly what I expect: a sine wave whose frequency is defined by the freq parameter. In particular, higher frequencies have shorter wavelengths (more cycles per second means shorter waves). When I create a sawtooth wave, the opposite seems to occur: higher frequencies result in longer waves. But that?s not

Hi, there is an interesting case when the FLAC encoder (using 1.2.0) is given simple waves. Simple waves means: I have a list of {frequency, duration, pause} tuples that define the monophonic tune. In other words, exactly one frequency is played at a time. This is the original dataset from 1989 (driving a PC speaker back then): $ ls -l ihold.sd -rw-r--r-- 1 jengelh users 20616 Aug 14 00:57

Hello, Do you know of any Windows software that is able to record in mono from the sound card's line input and at the same time convert to *.ogg? Ultimately, one without ogg conversion would be OK too provided it can record mono wave files. Important: the software should be able to record WITHOUT first creating an empty wave file with a set length (Adaptec's Spin Doctor versus Goldwave).

Ccing the maintainer if the tuneR package. Looks to me like sawtooth (and square) don't behave as expected when using xunit="samples". Workaround is to use xunit="time" instead: sawtooth(110,duration=1/100,samp.rate=sample_rate,xunit="time") I looked at the code but found it to be opaque. -- Sent from my phone. Please excuse my brevity. On November 12, 2017

Hello all, I am trying out the echo cancellation of Speex (1.2beta2) on Windows XP. I use the Windows Media API (i.e. waveInOpen, waveOutOpen etc) for playback and capture, but I cannot make echo cancellation works in my use case. Here is a description of what I tried: (the sound is in 8kHz and 16 bits per sample, the frame size is 240, tail_length is long enough to cover the far end to near end

On Nov 26 22:39:27, hans at stare.cz wrote: > ./test_replaygain.sh fails for me in tonegenerator(), saying: > > ./test_replaygain.sh[91]: mawk: not found > Testing FLAC replaygain 8000 (8000 x 1) ... -: ERROR: got partial sample > > Apparently, the tone-generating awk script does not work with > my system's awk, which is "awk version 20110810" as distributed

I have been attempting to estimate the periodic contribution of an effect to some data but have not been able to fit a sine wave within R. It would be nice to start by being able to fit a sine wave with an amplitude and frequency. x<-seq(0,20,by=0.5) y<-2*sin(2*pi*.5*x) #amplitude =2, frequency=0.5 # This failed to converge r<-nls(y ~ A*sin(2*pi*F*x), start=list(A = 1, F = 1),

Hey all, I can't seem to get the princurve package to produce correct results, even in the simplest cases. For example, if you just generate a 1 period noiseless sine wave, and ask for the principal curve and plot, the returned curve is clearly wrong (doesn't follow the sine wave). Here's my code: library(princurve) x <- runif(1000,0,2*pi); x <- cbind(x/(2*pi), sin(x)) fit1

Just noticed that our new server has a newer 80+ efficiency 865W PSU with PFC. So apparenty our existing 1500VA (865W) APC Back-UPS is not safe to use anymore. We apparently need a pure sine wave UPS. http://nam-en.apc.com/app/answers/detail/a_id/8883 I found an affordable 1500VA UPS by CyberPower, model CP1500PFCLCD, which has a pure sine wave output. Wondering if anyone has experience with

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)

Hello, I've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

Thank you for writing and pointing out that one can have WAVE files without getting the warning, and that your library does it. I found out Goldwave (Windows software) writes 24-bit WAVE files that do not get the warning. (Good for them!) However, the flac executable from the flac reference software apparently produces WAVE files that do get the warning. Example annotated log: Encode a 24-bit

Hello, I'm trying to generate a sine wave in R to fit my observations using the general formula: y=a*sin(b[x+h*pi)]+k where a = amplitude, b=period, h=phase shift, and k=vertical shift I want to use following translation to bring the sine function up onto the y-axis to range from 0-1, and this will place the wave on the x-axis from 0-pi/2. y=1/2sin(2[x+ 1/4*pi]) + 1/2 Additionally, I need

Thank you for you quick reply, Jean-Marc. I have just used a human voice to replace the sine wave, but the result is like a broken voice after tens of seconds. Do you have some sample voice in .sw that I can test with? I want to have one set or a few sets of sample voice that work for my understanding and debugging. This is the output from echo_diagnostic.m in my human voice test: Far end to

Hi, I wanted to play around with the echo canceller so I tried the testecho program. As speaker sound I fed it with a 440 Hz sine wave, and as mic with a 550 Hz + 440 Hz sine waves sound (but the 440 Hz component half the amplitude). I kind of expected to get a somewhat clean 550 Hz sine out, but it doesn't seem to work that way. Are there better samples I can use with testecho to see

The aplication that I am developing is like a sound recorder (goes directly to the soundboard and start capturing the sound and writing the file, using the directsound api). I'm already capable of writing to normal wav files but I wanted to encode the audio with speex to have smaller size (and still good quality). How can I write the wav files with speex content using the ACM codec? (like