User talk:Galeandrews

Go to the pharmacy for your daily viagra.

Testing of ExtendedImport April 2010
Tested with identical results in Unicode Release on Ubuntu and Windows 7.

Definite problems in red. Queries in brown.

Summary:


 * In out-of-box state (launch with initialised .cfg), unable to import any native formats via FFmpeg using OpenFile
 * State of override filter affects behaviour even when there are no rules
 * If rule states to use FFmpeg and override checked, you have to select the specific format to get the native importer, not just "All supported files", which "may" be unexpected given pre-ExtendedImport behaviour

1. Initialised .cfg to all but "NewPrefsInitialised=1"

Import > Audio a number of MP3s with filter set to FFmpeg: all files import using libmad 06:51:56: LastOpenType is FFmpeg-compatible files 06:51:57: OverrideExtendedImportByOpenFileDialogChoice is 0 06:51:57: File name is /home/gale/desktop/4.mp3 06:51:57: Mime type is * 06:51:57: Applying default rule 06:51:57: Inserting libsndfile 06:51:57: Inserting liboggvorbis 06:51:57: Inserting libflac 06:51:57: Inserting libmad at 0 06:51:57: Inserting lof 06:51:57: Inserting libav 06:51:57: Opening with libmad 06:51:57: Open succeeded

Import > Audio a number of WAVs with filter set to FFmpeg: all files import using libsndfile 06:59:53: LastOpenType is FFmpeg-compatible files 06:59:53: OverrideExtendedImportByOpenFileDialogChoice is 0 06:59:53: File name is /home/gale/desktop/344long2.wav 06:59:53: Mime type is * 06:59:53: Applying default rule 06:59:53: Inserting libsndfile 06:59:53: Inserting liboggvorbis 06:59:53: Inserting libflac 06:59:53: Inserting lof 06:59:53: Inserting libav 06:59:53: Opening with libsndfile 06:59:53: Open succeeded

Similarly, FLAC imports with libflac when filter set to FFmpeg.

2. Add *.wav rule that has FFmpeg moved to top; override rule is checked.

Drag in WAV and it comes in via FFmpeg, MP3 via libmad.

Import > Audio a WAV, filter set to "All files" and "All Supported files. File imports via FFmpeg:

05:55:12: LastOpenType is All files 05:55:12: OverrideExtendedImportByOpenFileDialogChoice is 1 05:55:12: File name is /home/gale/desktop/344long.wav 05:55:12: Mime type is * 05:55:12: Testing extensions 05:55:12: *.wav 05:55:12: Match! 05:55:12: Testing mime types 05:55:12: Match! 05:55:12: Complete match! 05:55:12: Inserting libav 05:55:12: Inserting libsndfile 05:55:12: Inserting liboggvorbis 05:55:12: Inserting libflac 05:55:12: Inserting libmad 05:55:12: Inserting lof 05:55:12: Opening with libav 05:55:12: Open succeeded

06:03:16: LastOpenType is All supported files 06:03:16: OverrideExtendedImportByOpenFileDialogChoice is 1 06:03:16: File name is /home/gale/desktop/344long.wav 06:03:16: Mime type is * 06:03:16: Testing extensions 06:03:16: *.wav 06:03:16: Match! 06:03:16: Testing mime types 06:03:16: Match! 06:03:16: Complete match! 06:03:16: Inserting libav 06:03:16: Inserting libsndfile 06:03:16: Inserting liboggvorbis 06:03:16: Inserting libflac 06:03:16: Inserting libmad 06:03:16: Inserting lof 06:03:16: Opening with libav 06:03:16: Open succeeded

Import > Audio a WAV with filter set to WAV. File imports via libsndfile.

3. Add *.MP3 rule to use FFmpeg.

Drag in an MP3 and it uses FFmpeg

Import > Audio an MP3 with filter set to "All files" and "All supported files". File imports via FFmpeg. 06:14:50: LastOpenType is All files 06:14:50: OverrideExtendedImportByOpenFileDialogChoice is 1 06:14:50: File name is /home/gale/desktop/344long.mp3 06:14:50: Mime type is * 06:14:50: Testing extensions 06:14:50: *.wav 06:14:50: Not testing mime types 06:14:50: Testing extensions 06:14:50: *.mp3 06:14:50: Match! 06:14:50: Testing mime types 06:14:50: Match! 06:14:50: Complete match! 06:14:50: Inserting libav 06:14:50: Inserting libsndfile 06:14:50: Inserting liboggvorbis 06:14:50: Inserting libflac 06:14:50: Inserting libmad 06:14:50: Inserting lof 06:14:50: Opening with libav 06:14:50: Open succeeded 06:14:50: Stream 0 start_time = 0, that would be 0.000000 milliseconds.

Import > Audio an MP3 with filter set to MP3. File imports via libmad.

4. Uncheck filter override.

Drag in WAV and comes in via FFmpeg, drag MP3 and comes in via FFmpeg Import > Audio an MP3, uses FFmpeg with all possible filters Import > Audio a WAV uses FFmpeg with all possible filters Import > Audio a FLAC, uses libflac

5. Delete all rules, override checked

Import > Audio a WAV with FFmpeg filter, uses FFmpeg

Import > Audio an MP3 with FFmpeg filter, uses libmad

10:18:46: LastOpenType is FFmpeg-compatible files 10:18:46: OverrideExtendedImportByOpenFileDialogChoice is 1 10:18:46: Inserting libav 10:18:46: File name is f:\abba\lay_all_your_love_on_me_e_rotic.mp3 10:18:46: Mime type is * 10:18:46: Applying default rule 10:18:46: Inserting libsndfile 10:18:46: Inserting liboggvorbis 10:18:46: Inserting libflac 10:18:46: Inserting libmad at 0 10:18:46: Inserting lof 10:18:46: Opening with libmad 10:18:46: Open succeeded

Drag in an MP3 or WAV (filter is FFmpeg), imports via libmad or libsndfile. Probably OK, behaviour is as it was before ExtendedImport?

10:22:38: LastOpenType is 10:22:38: OverrideExtendedImportByOpenFileDialogChoice is 1 10:22:38: File name is f:\abba\chess misc\someone else's story ashleigh gray.wav 10:22:38: Mime type is * 10:22:38: Applying default rule 10:22:38: Inserting libsndfile 10:22:38: Inserting liboggvorbis 10:22:38: Inserting libflac 10:22:38: Inserting lof 10:22:38: Inserting libav 10:22:38: Opening with libsndfile 10:22:38: Open succeeded

6. Add *.wav and *.mp3 rules, behaviour as before; delete all rules, override unchecked.

Import > Audio a WAV with FFmpeg filter; uses libsndfile 06:36:29: LastOpenType is FFmpeg-compatible files 06:36:29: OverrideExtendedImportByOpenFileDialogChoice is 0 06:36:29: File name is /home/gale/desktop/344long.wav 06:36:29: Mime type is * 06:36:29: Applying default rule 06:36:29: Inserting libsndfile 06:36:29: Inserting liboggvorbis 06:36:29: Inserting libflac 06:36:29: Inserting lof 06:36:29: Inserting libav 06:36:29: Opening with libsndfile 06:36:29: Open succeeded

Import > Audio an MP3 with FFmpeg filter; uses libmad 10:33:26: LastOpenType is FFmpeg-compatible files 10:33:26: OverrideExtendedImportByOpenFileDialogChoice is 0 10:33:26: File name is f:\abba\you_owe_me_one.mp3 10:33:26: Mime type is * 10:33:26: Applying default rule 10:33:26: Inserting libsndfile 10:33:26: Inserting liboggvorbis 10:33:26: Inserting libflac 10:33:26: Inserting libmad at 0 10:33:26: Inserting lof 10:33:26: Inserting libav 10:33:26: Opening with libmad 10:33:26: Open succeeded

Text upload
You can either left-click to view this dummy patch file in your browser (in FF anyway), or right-click to download it:
 * [[Media:scroll.txt|scroll patch]]

(turned off text upload for now in case any security considerations need to be discussed)

Definition of the decibel scale
When displaying the waveform in decibels, values range from 0 to -Here is information about decibel levels and perceived volume change. Here’s the quick read info, with supporting documentation below.

* 3dB = twice the power (note + 3db volume increase requires double the power but        does not double the amplitude!) * 6dB = twice the amplitude * ~10dB = twice the perceived volume * Adding up two 12dB noise sources will get you, on average, 15dB (which will not sound twice as loud)

My newbie question is what are the y-axis units -- decibels below what reference?

dB is a ratio to the maximum signal, expressed in logarithmic form. Thus, -36 dB means that the amplitude of your sample is 10^(-36/20) = 0.016 of maximum scale (0 dB = 1). You probably discovered already that the scale can be adjusted in Audacity Preferences, Interface tab.

OK, but in this case, a ratio of what to what? For example, in working with WiFi radio power one typically uses dBm -- decibels relative to a milliwatt of power. What is being shown here?

The ratio between the amplitude of the current signal and the maximum possible amplitude. I don't know if i can explain it properly but i'll give it try...

DB is a ratio .sort of a arbritrarily choosen point... .. a ratio comparing the intensity of the sound to a reference level! if the intensity of the is equal to the reference level then that point is called 0db .....

The confusing part sometimes is understanding what is the 0 point... and it really is determined by what your specifically doing and measuring... because it can be different ie: if your recording or if your amplying or if your measure voltage or rf energy or sound intensity... if your looking at the 0 point being nothing or the 0 point be the max of something .. or something in between .. smile... that the bad part....smile...

for example.... if measuring sound intensity the 0 point is at that point where there no sound... and as sound is heard it will be be measure in terms of +db rating for example a jack hammer running, when measured, will have a intensity of appx +100 db's ... in reference to that ) 0 db where 0db is no sound at all..

or if your sending a signal... well the max signal strength may be considered at 0 db and any thing less than that would be a -xdb ... for example... if you send only half of the maximum signal strenth .. that would equate to appx -3 db......... compared to 0db

now that's only true as a measurement of power .... for voltage and sound pressure amplitude, - 6dB is half as loud (twice as quiet), -12 db is four times quieter -20 db is ten times quieter and -40 dB 100 times quieter

now what does this mean for programs like audacity...

when it come to recording..... 0db is considered the max signal strength before distortion/clipping start occurring ..!! this 0 db is going to be determined by your sound card... what it's circuitry can handle!!

if your amplyfying a signal in your audacity... the 0 db reference is at point where there is no sound signal and everything above that is a indication of signal strength in reference to 0 sound!

now I mentioned that 0db is a arbitrary point.... and it depends on what one ones want it to represent weather it represent a max signal or zero signal... however it can represent something betweend for example in your audacity.... on your wav form... if you notice the 0db ref line is in the middle of the wav form ... it showing the signal strenth above and below the 0 reference line... so in this case the zero is sort of a null point... i am not going to get into any deeper on this smile... but just to say sometimes in music the 0 reference actually is riding at higher point and some you need to bring the music fullyly to a zero refernce there even function in audacity for that.... but that enought of that.. smile..

well here some links to start off with...

http://science.howstuffworks.com/framed.h…?parent=questio…://arts.ucsc.edu/ems/music/tech_background/TE-06/teces_06.html

http://www.phys.unsw.edu.au/~jw/dB.html

http://searchsmb.techtarget.com/sDefinition/0,,sid44_gci21…

http://www.mmproductions.co.uk/faqgen1.html

http://trace.wisc.edu/docs/2004-About-dB/

http://www.kodachrome.org/salt/sunderst.htm

http://www.audiomasterclass.com/arc.cfm?a=what-is-the-difference-between-level-volume-and-loudness

http://www.conradaskland.com/blog/2008/10/decibel-levels-and-perceived-volume-change/

Nice links! The first reference tells that the reference level for sound pressure is a very low value of 0.02 mPa. Indeed it cannot be zero sound, because you can't calculate a ration to 0: Signal/0 = not determined. Thus, in audio, a very low sound that (some) people still can hear is used as a reference.

that right about the sound pressure.. but keep in mind that 0db is arbitrary value and can be giving any value depending on what is to be measured.... for sound pressure that .02 value is assigned based on standards, however if your looking a circutry, a rf device, etc etc etc.... the reference value will be different...!! smile....

What is difference between dB SPL and dB(A) SPL?

The following is from the Acoustics FAQ. A sound level meter that measures the sound pressure level with a "flat" response will indicate the strength of low frequency sound with the same emphasis as higher frequency sounds. Yet our ear perceives low frequency sound to be of less loudness that higher frequency sound. The eardrum- stapes-circular window system behaves like a mechanical transformer with a finite pass band. In EE parlance, the "3 dB" rollover frequencies are approximately 500 Hz on the low end and 8 kHz on the high end. By using an electronic filter of attenuation equal to that apparently offered by the human ear for sound each frequency (the 40-phon response curve), the sound level meter will now report a numerical value proportional to the human perception of the strength of that sound independent of frequency. Section 8.2 shows a table of these weightings.

Unfortunately, human perception of loudness vis-à-vis frequency changes with loudness. When sound is very loud - 100 dB or more, the perception of loudness is more consistent across the audible frequency band. "B" and "C" Weightings reflect this trend. "B" Weighting is now little-used, but C-Weighting has achieved prominence in evaluating annoying community noises such as low frequency sound emitted by artillery fire and outdoor rock concerts. C-Weighting is also tabulated in 8.2.

The first electrical sound meter was reported by George W Pierce in Proceedings of the American Academy of Arts and Sciences, v 43 (1907-8) A couple of decades later the switch from horse-drawn vehicles to automobiles in cities led to large changes in the background noise climate. The advent of "talkies" - film sound - was a big stimulus to sound meter patents of the time, but there was still no standard method of sound measurement. "Noise" (unwanted sound) became a public issue.

The first tentative standard for sound level meters (Z24.3) was published by the American Standards Association in 1936, sponsored by the Acoustical Society of America. The tentative standard shows two frequency weighting curves "A" and "B" which were modeled on the response of the human ear to low and high levels of sound respectively.

With the coming of the Walsh-Healy act in 1969, the A-Weighting of sound was defacto presumed to be the "appropriate" weighting to represent sound level as a single number (rather than as a spectrum). With the advent of US FAA and US EPA interests in the '70's, the dBA metric was also adapted by them. (Along with the dBA metric has come an associated shortfall in precision in accurately representing the capacity of a given sound to produce hearing loss and the capacity to create annoyance.)

[Editor's Note: A single number metric such as dBA is more easily understood by legal and administrative officials, so that promulgation, enforcement and administrative criteria and actions are understandable by more parties, often at the expense of a more precise comprehension and engineering action capability. For instance, enforcement may be on a dBA basis, but noise control design demands the octave-band or even third-octave band spectral data metric.]

The most commonly referenced weighting is "A-Weighting" dB(A), which is similar to that originally defined as Curve "A" in the 1936 standard. "C-Weighting" dB(C), which is used occasionally, has a relatively flat response. ""U-Weighting"" is a recent weighting which is used for measuring audible sound in the presence of ultrasound, and can be combined with A-Weighting to give AU-Weighting. The A-Weighting formula is given in section 8 of this FAQ file.

In addition to frequency weighting, sound pressure level measurement can be time-weighted as the "Fast", "Slow" or "Impulse" response. Measurements of sound pressure level with A-Weighting and fast response are also known as the "sound level".

Many modern sound level meters can measure the average sound energy over a given time. this metric is called the "equivalent continuous sound level" (L sub eq). More recently, it has become customary in some circles to presume that this sound measurement was A-Weighted if no weighting descriptor is listed.

image uploads and what I tried to do
Hi Gale,

thanks for stopping by on my user page. I wanted to link the images for a user pages template to make it a bit prettier. You can see it in action at Wikipedia at http :// en dot wikipedia dot org / wiki / User:Cumbrowski (remove the spaces and "dot" with "." ... sorry but the inability to use any links on the user pages is responsible for the inconvenience :) )

Cheers! --roy&lt;sac&gt; Talk!  .oOo. 05:24, 22 November 2008 (PST)

Just saw other links on your talk page.. lets try it too. http://en.wikipedia.org/wiki/User:Cumbrowski
 * thanks for the response and the tips Gale. I left a more detailed response for you at my User Talk page --roy&lt;sac&gt; Talk!  .oOo. 18:22, 26 November 2008 (PST)

Page anchors
Jeff Q 16:58, 1 June 2009 (PDT): I noticed that you added a  anchor before the heading "Troubleshooting MP3 export problems" in Lame Installation. You don't really need one; MediaWiki automatically generates anchors for H2-level headings like that one. This text:



takes you right to that section. Try it here:


 * Lame Installation

You can also add a link label like this:



which looks and works like this:


 * troubleshoot LAME MP3 export problems

Hope this helps.


 * Gale 02 Jun 09: Thanks Jeff, everything you say is of course correct. But the reason we add  anchors that appear superfluous in terms of the Wiki is to link to these sections in user support responses - the URLs that Wiki headings produce (though fine for TOCs) are often simply too long otherwise.   And just to add a tip of my own in case any one else is reading - if you are linking to a heading within the page, you can just use the   without the root URL.  - Gale


 * Jeff Q 11:36, 2 June 2009 (PDT): Thanks for the Audacity Wiki tip, and sorry for the bad assumption. It only occurred to me after I posted that there might be some special reason here for your markup. D'oh! And I'll bet you suspected I might know the #-only tip too, but I appreciate it anyway – I've learned quite a lot of wiki-lore just this way.


 * As long as we're sharing stuff we probably already know &#9786;, you can abbreviate the extra anchor by making it an empty element like so: . (Or  , with the extra space, if you want to avoid confusing very old browsers.) And   works as well as  , if there's ever a place where   causes problems. Apologies again if this info is also redundant.

pt/br?
não sei se voçê entende português, porque só esta wiki é pt/br quem formulou essa regra foi votada, idiomas separados por país e não por etnia? estranho toda wiki que eu edito o português tanto é para pt Brasil pt Portugal pt Angola e etc queria entender o porque desse egocentrismo.Rautopia 08:18, 13 June 2009 (PDT)

Sphinx Search
I noticed your comments regarding using the SphinxSearch on this wiki. I am now working on adding new features to it, so feel free to let me know what would make our extension more useful to you - in case you are still considering using it. All the best! Svemir 22:06, 14 December 2009 (CST)

(OGG) Thanks for replying :)

 * I'll make the request to support less than (-q0), however, I couldn't find the place to put my request.  After searching so much, I found a standalone encoder which makes (-q-1) & (-q-2) OGG Vorbis files: Oggenc2.85using aoTuVb5.7. I only tried to convert from FLAC to OGG with it.
 * You asked me if I were asking for an automatically lower setting for mono audio, well, it would be a very good idea, because I always feel that (-q2) in mono corresponds to (-q3) in stereo. It would be indisputable if this feature were used if we want to or not used if we didn't want it. It would be nice to be adjustable, so that, for example, I may adjust that Audacity automatically chooses a less-by-one-step-quality for the export quality for mono. Or, I may choose that the quality be reduced by two steps for mono.
 * About what I named in my opinion as efficient quality settings for stereo: I don't have citation for that. It's my own opinion. I feel that (-q10) would be chosen if we want to make perceptually lossless audio files. I even feel that 384Kbps MP2 isn't perceptually lossless.  I assume that users would only need such high quality settings such as (-q10) when they want a perceptually lossless audio file.   (-q9) is not perceptually lossless & doesn't really sound better than (-q8).   (-q4) doesn't really sound better than (-q3), if the effect, that is used to remove the voice of the singer (the sound in both left & right channels), were added on a (-q4) file, it would sound worse than in 128kbps MP3 with the same effect :(
 * When I listen to stereo files less than (-q2), it feels that the sound has become less stereo (I don't know how! I feel that the stereo separation is less. It is especially audible in loud music, example: Hard Rock). So, if someone wants a small file-size/bit-rate, then he would rather choose (-q0) & not (-q1).


 * In the end, I wasn't suggesting to remove -q (0), (1), (4) and (9). I was only writing my opinion. Thanks again :) --Dudi 23:14, 8 February 2010 (CST)

Sortable feature request
I've just made a try to help improving the sorting of the feature requests and I'd like to have your opinion. If it's ok for you, I'll be able to adapt the actual feature request page. There will be no need for the highest rated section. --Djiboun 08:39, 2 October 2010 (UTC)