search for: decibels

...window", taper.=taper(type="hamming",n.sample=length(data))) that produces negative values, the decibel() function expectantly throws error. f=attr(vysl,"frequency")[-1] s=decibel(as.vector(S)[-1]) Error in decibel(as.vector(vysl)[-1]) : cannot convert negative values to decibels Is it possible to use some other function than decibel, or how else can estimate spec.density with tapers from this package? Thank you very much. [[alternative HTML version deleted]]

Hello, I've read in a thread from this mailing list ( http://lists.xiph.org/pipermail/vorbis-dev/2009-September/020019.html) that the usage of FLOOR1_fromdB_LOOKUP table is to convert Floor1 values from a dB scale to a linear scale. In a dB scale, given a reference value, the other dB values are obtained from it, normally dB_value = 10log(p1/p0), where p0 is the reference value. Supposing p0

I'm considering patching my copy of ogg123 to support vorbisgain tags. Is there any interest in folding this sort of patch into the main line ogg123? Has anyone done this already (or to another command line ogg player)? John --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 There has been a lot of discussion on these two mailing lists about the upcoming EoL of FreeBSD 4.x which I mentioned in my email entitled "HEADS UP: FreeBSD 5.3, 5.4, 6.0 EoLs coming soon". Now that everybody (hopefully) has had their say, I'd like to offer some background and explanation. The concept of "security branches"

Hi. Since I'm new to this mailing list, I don't know whether this was discussed in this mailing list, or if it is already fixed. Anyway, the problem is that all audio encoders out there add a small amount of silence to the start/end of the encoded file. This is usually a small amount, around 0.03 seconds. Most of the time it goes unnoticed, BUT when you have tracks that flow into one

Hi, Attached is a patch which enables FFTW3 to be used by libspeexdsp. Using SSE, the FFT code is more than twice as fast, and it's 50% faster even without SSE. To enable the library, you'll need to call configure with '--with-gpl-fftw3'. Based on the benchmarking we did, this also changes the default FFT to smallft for floating point implementations when not using FFTW3.

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

...ons for leaving the message will be skipped. * 'u' the "unavailable" greeting will be played. * 'b' the "busy" greeting will be played. * 'g(#)' the specified amount of gain will be requested during message recording (units are whole-number decibels (dB)) In my case, the asterisk box is located about 7db from the central office. As noted in bug 2023 (and 2022), calls from an outside pstn line coming into asterisk incure a 7db pstn loss (which can't be adjusted for with rxgain and txgain as changing those values to something reasonable ge...

I've been experimenting with the ideas of Replay Gain[1] and find that ogg123 doesn't have a way of specifying the scaling applied to replayed samples (like -f in mpg123). Looking at libvorbisfile, I see no function exactly matching this possibly desirable behaviour. ov_read() scales by either 128 (byte output) or 32768 (word output), but there's nothing in between. ov_read_float()

...> prehaps 24 bit, was the maximum we could do. The only place I've ever heard of 96 bit anything is in image processing where R, G and B were each encoded as a float. If the RGB explanation is not what you are thinking about, do you realise that 96 bits gives a dynamic range of over 500 decibels? That means that if the largest sample corresponds to 1000 volts, the smallest sample will correspond to 5e-26 volts. I think you'll find that this is less than the voltage on an electron which is approx. 1e-19 volts. > And yet, the issue was raised that 64-bit audio samples > are nesse...

...FLAC to be around 33% of the 24-bit WAV if it actually only had 16- bit samples, instead of the 40% that you're seeing. But I suppose that's always possible. Brian Willoughby Sound Consulting P.S. I'm still curious where Nicholas came up with the 14% value. Is that based on decibels, bits, or some other metric I haven't thought of?

Hi all, How can i make voicemailmain application to not ask user mailbox. it should only ask for password. In the prev versions i used the 'u' option for this purpose but now its gone. So is there still a way to do this? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...example: On title, send always actual Frequency of receiver, so if this change, clients can view the changes. This updates on metadata must be refreshed too much, on every second of stream will be acceptable. On artist maybe the actual mode of receiver (AM/FM...) or incoming signal station (decibels). I know that maybe metadata was not created for this purpose, but for my project, this solution is perfect, because only one channel of data is used. FAQ tells that there is some way of multiplexing data in the OGG container, but I don't know if this multiplexing is done for every fra...

...char *cbuffer ; f1 = 1000 ; // frequency in Hertz f2 = 1.5 * f1 ; // perfect fifth higher ph1 = ph2 = 0 ; phInc1 = ( 2 * M_PI * f1 ) / sr ; // phase increments phInc2 = ( 2 * M_PI * f2 ) / sr ; db = 90 ; // amplitude in decibels A = pow(10.0, db/20.0) / 2 ; // linear amplitude frames = dur * sr ; *nBytes = frames * nchan * sizeof(short) ; sbuffer = (short *)malloc(*nBytes) ; if ( nchan == 1 ) { // mono sound for ( i = 0 ; i < frames ; i++ ) { sbuffer[i] = (shor...

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

...the 24-bit WAV if it actually only had 16-bit samples, instead of the 40% > that you're seeing. But I suppose that's always possible. > > Brian Willoughby > Sound Consulting > > P.S. I'm still curious where Nicholas came up with the 14% value. Is that > based on decibels, bits, or some other metric I haven't thought of? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20101202/07207870/attachment.htm

...standard for that? It's not about what's now, it's about what could be, and nobody has been able to make good predictions, except your logic here: > If the RGB explanation is not what you are thinking about, do > you realise that 96 bits gives a dynamic range of over 500 > decibels? That means that if the largest sample corresponds to > 1000 volts, the smallest sample will correspond to 5e-26 volts. > I think you'll find that this is less than the voltage on an > electron which is approx. 1e-19 volts. This is an excellent argument. It's still not a cap, sin...

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Hi, While testing ReplayGain (so it could be related to a bug), I noticed the following gain comments for a file: RG_PEAK=1.71580 RG_RADIO=-7.91 dB RG_AUDIOPHILE=-6.72 dB I've never seen such a large peak. Not that I've looked much, nor have I analyzed the file further. I just thought I should mention it. :) The file is the track Board Burner by Mixmaster Mike, available at:

Hi all, I just came across the following page: http://r3mix.net In the "News" section there is a fairly negative critique of Vorbis; especially it is mentioned that Vorbis "has terrible masking problems". There is also a Vorbis-encoded frequency sweep which shows strong deficiencies at high frequencies, but I suppose this is due to the quite low bitrate of the distributed