similar to: a way restrict encoding sound volume

Dear Prof Ripley, First of all, unless you are an english professor, then I do not think you have any business policing language. I'm still very much a student, both in R, and regarding signal analysis. My competence on the subject as compared too your own level of expertise, or my spelling for that matter, may be a contension for you, but it would have been better had you kept that opinion

Dear list, I apologies first for my English, hope you will understand well my question. I am working on 1/2 hour piezometric data, time unit is second. They present daily oscillation when using the spectrum() function. What I am really interested in, is to find the amplitude corresponding to this oscillation. I work with a college using Matlab, and although we apply the same methodology, our

Hi, here is a practical problem we would like to solve. In a pneumatic post the acceleration of the capsule is measured and plotted over time. From the graph achieved we would like to derive some kind of statistic value that describes the stress the capsule, or what is in it, is exhibited to. The amount of stress introduced to the capsule will probably depend on two things, the maximum

Hello, I have two questions concerning floor1 encoding/decoding. First I'll ask about the FLOOR1_fromdB_LOOKUP table: what is it's purpose? Is it to convert the amplitude differences between [floor1_Y] values to a dB scale? And, if I'm right with that, here comes the 2nd question: when render_line0 is used to encode floor1, then floor1_inverse2 must be used in decode (in order to

Hi, It is better to replace this line in function filterbank_new: max_mel = toBARK(EXTRACT16(MULT16_16_Q15(QCONST16(.5f,15),sampling))); to max_mel = toBARK(EXTRACT16(sampling/2)); It gives the same but it seems to be faster and avoids overflow on 44100 kHz that prevents denoiser to process 44100 streams. (Yes I know that Speex should not pack 44100 streams but it does now and I use it). Best

Hi All, Here is the problem: I'm trying to generate a number of Fourier Descriptors figures for an experiment. All I need is that they are created within a loop and saved with sequential names (e.g., s1_1.png, s1_2.png etc..) in my directory. I created a nested loop with a counter for the different amplitudes for the actual shapes and a counter for the file names. This script: *count

There's a new module in CVS called 'masktest'. I spent the last few weeks writing this program. It can measure masking based on input from people. We need it to obtain better masking curves than we have right now (from Ehmer...). It's not finished yet, but you can get a feeling of what it's supposed to do. It will measure the masking between tone and tone, noise /tone,

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

Dear list, What on earth is spec.pgram() normalized too? If you would like to skip my proof as to why it's not normed too the mean squared or sum squared amplitude of the discrete function a[], feel free too skip the rest of the message. If it is, but you know why it's not exact in spec.pgram() when it should be, skip the rest of this message. The issue I refer herein refers only too a

Dear list, I' m studying FLAC performance, and I'd like to know how much compression can be achieved for different audio files. 1) It seems that for nontonal sound (wideband noise), the compression factor is better than for compound sound (tones + nontonal components), which is typically 2. The reason for this result could be the following: the LPC filter is more suitable for

Hi, I'm trying to use a denoiser on a wince with a FIXED_POINT defined. Denoiser works OK - it removes the noise, but then it unacceptable hurts a voice. Here is a code that I use: #define TEST_DENOISE_SAMPLES 2000 void test_denoise() { FILE *fin; FILE *fout; spx_int32_t rate=0; int chan=1; int fmt=16; int denoise_enabled = 1; SpeexPreprocessState *preprocess;

Hi, I have some files in mp3 in my Asterisk but when I play it the volume is lo= wer than wav files. Both the files (wav and mp3) are encoded with the same = amplitude. In anothers players the audio volume of these files are equal. Can I fix this diference between volume of mp3 and wav file? Thanks Veja quais s?o os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com

Dear helpers I constructed a data frame with this structure > str(dados1) `data.frame': 485 obs. of 16 variables: $ Emissor : int 1 1 1 1 1 1 1 1 1 1 ... $ Marisca.Rio : int 1 1 1 1 1 1 1 1 1 1 ... $ Per?odo : int 1 1 1 1 1 1 1 1 1 1 ... $ Reproducao : int 3 3 3 3 3 3 3 3 3 3 ... $ Estacao : int 2 2 2 2 2 2 2 2 2 2 ... $ X30cm : int

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to

Hi! I'm learning mgcv, and reading Simon Wood's book on GAMs, as recommended to me earlier by some folks on this list. I've run into a question to which I can't find the answer in his book, so I'm hoping somebody here knows. My outcome variable is binary, so I'm doing a binomial fit with gam(). I have five independent variables, all continuous, all uniformly

Hello, In doing a series of non-linear estimations of a function which is a sum of a varying number of sinusoids, I would like to "autogenerate" the arguments needed by nls() depending on that number. For example, when there are two sinusoids: > nls( y ~ mu + A1 * cos(2*pi*f1*x - P1) + A2 * cos(2*pi*f2*x - P2), data = some.xy.data, start = list( mu=some.value0,

Dear R help, I have some readings in three dimensions (x, y, z) and an amplitude for each. I'd like to visualize the data using ggplot, using tile plots, as I have some additional point data I would like to eventually overlay on the tile plots. I would like to subset the data by sections, slices if you will, in the z dimension, and plot the data for that slice. I can do all of this, but am

Please look at what I see in your code below (run-on code mush) to understand part of why it is important for you to send your email as plain text as the Posting Guide indicates. You might find [1] helpful. [1] https://wiki.openstack.org/wiki/MailingListEtiquette -- Sent from my phone. Please excuse my brevity. On June 25, 2017 2:42:26 PM EDT, Alaios via R-help <r-help at r-project.org>

Hi all,I had a question last week on asking for a function that will help me draw three different circles on x,y,z axis based on polar coordinates (Each X,Y,Z circle are coming from three independent measurements of 1-360 polar coordinates). It turned out that there ?is no such function in R and thus I am trying to write my own piece of code that hopefully I will be able to share. I have spent

Hi! I'm sending this message to both lame and vorbis developers, since my concerns apply to both ( and in case of vorbis it probably applies to the more advanced ambisonic modes the X, Y and Z parts ) Recently I did some thinking about M/S encoding and wondered if the same psychoaccoustic model is applied to the S ( and M ) channel as to normal L and R channels. My concern is that different