similar to: A masking test program

Hi! Can someone say to me, how the following values are assigned? I assume that it concerns different frequencies, but which? Excerpt out: ?masking.h? .. static float tonemasks[P_BANDS][6][EHMER_MAX]={ /* 62.5 Hz */ {{ -60, -60, -60, -60, -60, -60, -60, -60, -60, -60, -60, -60, -62, -62, -65, -73, -69, -68, -68, -67, -70, -70, -72, -74,

Dear List-Members, I would like to draw the amplitudes of different frequencies from a time series as shown in the attached figure. Does anybody has an idea how to do it? Best wishes Thomas -------------- next part -------------- A non-text attachment was scrubbed... Name: milankovich.pdf Type: application/pdf Size: 93605 bytes Desc: not available Url :

Hi, I need a psychoacoustics model for some experiments and would like to use the one included in vorbis. Is there any documentation about how to use it? Looks like the relevent code is in psy.c, but I can't go further. Also I would like to know whether I can be of any help in vorbis. I have some knowledge in speech coding (I doing my master at the University of Sherbrooke's speech

Is there a way to make an FXTable''s column widths (and row heights) unchangeable by the user? By experimentation it looks to me like leaving out TABLE_COL_SIZABLE and TABLE_ROW_SIZABLE from the options field in FXTable.new() has no effect. I can still drag the intersection between two column heading buttons to change the column width. The code I''m using to test this is

I am having a little problem in relation to sending output to the R Console window, when a function is called by pushing a button in a Tk window. Actually there seem to be two components to the problem: -- the R prompt does not re-appear until <return> is pressed -- text output to the R Console window *disappears* after <return> is pressed The following, in R 1.1.1 under

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 folks, Since the ASA nixed the idea of posting those papers on the web, I distilled out the raw masking curves. I also added in a few of the raw masking curves as published in "Paramaetric Bit Allocation in a Perceptual Audio Coder"; this paper only mentions in passing that Dolby distilled their masking data for AC-3 research from both the Ehmer papers I already mentioned and later

This works really well for white noise reduction. However what I've noticed was the amplitudes of normal speech samples also get reduced. Is this something by design, or is there a way to automatically recover the original speech sample volumes ? <p>Thanks. <p>Tongbiao <p>-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@hermes.usherb.ca] Sent:

Hi everyone, after having an extensive look at the reference decoder as well as the specification documents, I'd like to point out what I think to be some minor omissions/flaws in the latter, and suggestions for correcting them. --- #1: in vorbis-spec-floor1.html, packet decode: Between 14) and 15) a crucial step is missing. Please insert: 14a) [cval] >>= [cbits] ---- #2: in

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 vorbis developer! I'm not very experienced in informatics, cause i'm only an audio engineer and sorry for my broken english but i have an idea... After the filterbank the maximum aplitude in each band decides about the nessesary bits, then the psychoacoustic model decides how much of the signal can be truncated without hearable noise. What do you think about quantising the

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

When I empty and refill an FXTable multiple times with a large data set (10,000 rows x 1 column in the example below, around 1000 rows x 6 columns in a more complicated app), the application eventually fails with a segmentation fault. The number of repetitions required to produce the segfault is rather random, ranging from just a couple reps to 15 or 20 in some trials. The program disappears

I tested Vorbis encoder - beta3 version, and here are my thoughts: - In comparison to beta2, subtle high-frequency artifacts seem to be gone (though they were small in beta2). Good job there! :-) - Velvet.wav also sounds better, but transparent quality is reached at -b256+. - Horn.wav still sounds very sucky, mode -b256 gives ~100kbps (this is understandable because in this sample practically

You have the point. "I get very distorted speech when using libspeex in my application. What¡¯s wrong? .... If the input speech has an amplitude close to ¡¾2^15, it is possible that at decoding, the amplitude be a bit higher than that, causing clipping when saving as 16-bit PCM." This is one of FAQs in 'speex-manual' you wrote. I guess amplitudes close to 2^15 bother me. Do

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

I am getting the following errors when compiling SAMBA 2.2.4 on HP-UX 11.00. Anyone have any ideas on what's causing the problem? Using FLAGS = -O -I./popt -Iinclude -I./include -I./ubiqx -I./smbwrapper -D_HPUX_SOURCE -D_POSIX_SOURCE -D_LARGEFILE64_SOURCE -D_ALIGNMENT_REQUIRED=1 -D_MAX_ALIGNMENT=4 -DLOGFILEBASE="/opt/samba/var" -DCONFIGFILE="/opt/samba/etc/smb.conf"

I am getting the following link errors when compiling SAMBA 2.2.4 on HP-UX 11.00. Anyone have any ideas on what's causing the problem? Using FLAGS = -O -I./popt -Iinclude -I./include -I./ubiqx -I./smbwrapper -D_HPUX_SOURCE -D_POSIX_SOURCE -D_LARGEFILE64_SOURCE -D_ALIGNMENT_REQUIRED=1 -D_MAX_ALIGNMENT=4 -DLOGFILEBASE="/opt/samba/var"

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

Hi, I have an nxp data set of time series. For my final year project, I would like to partition this data set into a smaller number of units ( < n). Where each of the units contains time series that move closely together (i.e. in unison), and the "simmilarity" (in terms of occurence & amplitudes of peaks and troughs) between the segregated units is "low". I know