Audacity appears to be increasing the bit rate?

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Submariner
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Audacity appears to be increasing the bit rate?

Post by Submariner » Sun Jan 08, 2012 8:22 pm

When I ripped audio files in Audacity it appeared to be exporting at a higher quality!
Q1. Can anyone explain how this is? - I understand one can compress a file to a lower quality but how does it improve the quality without the data there? Any explanations much appreciated.
Q2. I suppose the next question is, if it's 'inventing' the data or whatever would I be better exporting in a lower quality? if so how does one do that, or does it not matter?
Q3. Is there a way to see all the audio data in Windows Media Player using Windows 7 like you can in iTunes Ref: a] Frequecy Sample Rate eg 44.1kHz or 48 kHz, b] sample size eg 16-bits; c] bitstream rate eg 1411 kbps ? I can only see the total file size and and bitstream rate eg 1411 kbps.
Q4. In light of the DVD Recorders specification [see below] am I right that the highest audio recording quality I can achieve on this recorder will be 256 kbps, FS 48kHz? i.e. it can playback a higher audio level than it can record?. Just trying to see if I can improve the quality of the wav file.
Any views appreciated
Peter

Brief Outline of what I did and the results:-
I recorded [in 'Video' format] a Vinyl LP on my Sony RDR-GX7 DVD Recorder [purchased in 2004], using the Line 4 L/R Audio Inputs] , when I played back the DVD-RW disc on this same recorder a bit rate of 256 kbps and a frequency sampling rate of 48kHz was shown on my AVR's 'Audio Input Stream' display panel. I think this was right [see below ref DVD Recorder's Specs].

However when I imported the stereo audio file [VOB] from the DVD disc into Audacity it showed 16-bit 48Khz, and the resultant .wav file exported by Audacity then had a frequency sampling rate of 44.1 and sample size 16-bit [so I assumed 44.1kHz in 16-bit format was correct, as that is what I had set as my Project Preferences].
Also when I looked at the bit rate of the file that was in Windows Media Player it just showed 1411 kbps. So I assume that is 44.1 kHz Frequency Sampling x 16-Bits x 2 channels = 1411kbps.
Also in iTunes the imported WAV file was 16-bits, 44.1kHz, 1411kbps].

The DVD Recorder's Specifications in the manual states [that produced this recording on a DVD-RW in Video format]:

Audio Recording Format /applicable Bit rate:
Dolby Digital/2Ch, 256 kbps.'
Video Recording Format:
MPEG Video
Strangely it says : 'Audio Characteristics (Playback) Frequency Response DVD (PCM 96kHz) : DVD (PCM 48kHz)'
It also says: 'This Recorder can NOT Play DVD Audio Discs'

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Re: Audacity appears to be increasing the bit rate?

Post by kozikowski » Sun Jan 08, 2012 11:08 pm

Audacity is not a good INFO tool. Every time we bring it up, we're told that it doesn't make any difference because the specifications are changed once the music is inside Audacity anyway.

Because of that, it's possible to edit an MP3 file by taking pieces out and export a larger file than you started.

You need to be careful with DVDs because they can and are likely to contain two different versions of the same show: Dolby Digital Surround which is a highly compressed and managed format and PCM which is straight uncompressed stereo WAV. If you start with one and end with the other, the new show, minute for minute, is going to be substantially larger than the old one, not to mention missing four of the six audio tracks.

Koz

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Re: Audacity appears to be increasing the bit rate?

Post by DVDdoug » Tue Jan 10, 2012 9:59 pm

When I ripped audio files in Audacity it appeared to be exporting at a higher quality!
Q1. Can anyone explain how this is? - I understand one can compress a file to a lower quality but how does it improve the quality without the data there? Any explanations much appreciated.
it's not improving the quality, it's just converting to a higher resolution format. If you copy a VHS tape or DVD to Blu-Ray, it's now in a high resolution format and there's more data on the Blu-ray than on the DVD, but of course the picture quality is not improved. Theoreticaly, there is quality loss due to the conversion, but you are unliklely to notice any difference.

With lossy compression, the "loss" happens during compression. So, if you open a 256kbps Dolby AC3 file, edit it, and re-save it as 256kbps AC3 (or 256 MP3), you are going through a 2nd lossy compression and there is (theoretically) additional quality loss. Even if you save at 320kbps, there is theoretical quality loss. (You may not hear any quality loss in either case.)
I suppose the next question is, if it's 'inventing' the data or whatever would I be better exporting in a lower quality? if so how does one do that
Ideally, you'd like to avoid any "extra" conversions. Conceptually, "upscaling" is done by interpolation, but the actual math is usually a bit more complicated. (When you decompress the Dolby Digital file to WAV, the math is much more complicated than interpolation.)

In your other post, you said you were making CDs. CDs are 44.1kHz, 16-bit, stereo, uncompressed. So, that would be a good format to use. If you de-compress the 256kbps Dolby Digital file to 1411kpbs WAV, of course the file is going to be about 5 times the size. The file gets decompressed when you play it anyway, so you're just doing it in advance... The only difference is that when you play the compressed file, it gets decompressed to 48kHz, and you'll be converting to 44.1khz for the CD.
i.e. it can playback a higher audio level than it can record?
Yes. All DVD players are required to play Dolby Digital up to 640kbps and uncompressed LPCM.
Just trying to see if I can improve the quality of the wav file.
It would be best if you could record to LPCM or WAV, but the weak link is the analog vinyl. (I wouldn't worry about the resolution of the WAV file). CD audio has more resolution than your original vinyl... i.e. You can probably hear surface noise between tracks on your records... It's usually not hard to tell if the record stops rotating, or if the stylus has been lifted from the groove. Some "audiophiles" claim that analog has infinite resolution, but the resolution is limited by noise (rather than by bit-depth). You don't hear any noise between tracks on a CD, because digital silence is true silence... If you do hear any noise between tracks on a CD, it's coming from the analog circuitry (or the CD was made from an analog source and not muted between tracks).

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Re: Audacity appears to be increasing the bit rate?

Post by Submariner » Wed Jan 11, 2012 2:41 pm

DVDDOUG
"Ideally, you'd like to avoid any "extra" conversions. Conceptually, "upscaling" is done by interpolation, but the actual math is usually a bit more complicated. (When you decompress the Dolby Digital file to WAV, the math is much more complicated than interpolation."

When I imported the VOB filess initially I set my preferences to 48kHz, my logic [probably flawed] was bring it in at the highest format you can then do the editing 'clicks and pops and splitting into tracks' then export the highest level format (as a Master if you like) and then Export again as a 44.1 kHz 16 bit (CD Master).
Q1. Would I therefore be better setting the project at 44.1 for the initial Import, edit it and Export the CD 44.1 as the main Master to avoid loss on multiple conversions?

I could buy a Pro-Ject Phonobox USB V. Advantages:
A) It might be quicker.i.e. just recording once to get a higher single .wav file (master). and Saves buying 50x DVD-R discs.
B) I could move the Turntable to a spare office bedroom and do record by record. - better for me, if these recording are pretty good.
C) The Project's specs say it can have Sampling rates of 32, 44,1 and 48kHz. But there is no physical way [buttons etc] of changing the setting. There is no spec for the sample size but it does say it has a Burr Brown A/D converter 16-Bit Delta Sigma. So should I assume 16-bit will be the highest it can go? i.e. 48kHz 16-bit (1536 kbps).
Q2. So if I set it to 44.1kHz 16-bit [as discussed above] would I be able to hear the difference? Would it be significant? I've got Dynaudio Accoustics Speakers [proffessional range - usually sold to Mastering Studio's] Amps have a 0.015 THD - as a guideline to the system quality.

Q3.Re setting the sampling rate and size is this effectively done by setting 44.1kHz as the Audacity Project Preference? or is it something to do with the soundcard settings ???? er completely lost here, as I have a laptop with no soundcard.
When I right clicked on my tooltray speaker icon, Sounds, Recording devices, all I can see is a Microphone! Hopefully if/when I the phono preamp/USB gadget is plugged in being USB it will appear there majically. It says Realtec High Definition Audio under the Mic icon. It also says Controller: Realtec High Definition Audio. I wonder if the this controller will be used to run the USB Phono box [I think you've guessed I havent a clue what this is all about]
Looking at Properties > Advanced > 'Select the sound rate and bit depth to be used when running in shared mode' panel is set to 16-bit 48kHz but this has a drop dowm menu going up to 24-Bit 96kHz.
Q4. So will I have to set the bit depth anf Frequency sample rate here as well as set it in Audacitly's project preferences?

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Re: Audacity appears to be increasing the bit rate?

Post by waxcylinder » Wed Jan 11, 2012 6:35 pm

I did move my TT to my spare-room/office close by my PC, jus as you are thinking of doing.

My recording chain is TT=>ART DJ Pre-11 pre-amp=>Edirol UA-1EX (USB soundcard)=>SkyTronic passive switch=>PC(XP-HE)

The passive switch lets me switch easily between 3 input sources (TT/ cassette deck/FM tuner) - it has a very low noise floor.

I record with the Edirol set to 16-bit (my understanding is that that is the most that Audacity can handle, down to the audio i/f s/w that it uses). However I work with Audacity set to 44.1kHz 32-bit float to give good headroom for editing. After all processing I export a set of WAVs, one per LP track at 16-bit PCM stereo. I then backup two copies of these WAVs to two separate 1TB USB disks (I don't want to lose all the valuabel work. The files are then deployed by loading into iTunes and converting to AAC at 256 VBR and/or I burn CDs with the WAVs.

The output is played on high-end kit: Rega Planet CD deck to QUAD ESL-57 electrostatic speakers and QUAD pre and power amps - the resulting sound is extraordinarily good (even the AACs from itunes in the iPod dock connected to the QUAD kit).

If I was being purist I guess I would also archive the original capture by exporting 32-bit WAV files and saving those in case future digital processing improves - but I am not an archivist and I'm getting too old to want to do all the re-processing - I want to spend my time listening to the music.

One thing I do that both speeds up my workflow and produces better results for click/pop removal is to use Brian Davies' excellent ClickRepair software (costs Aus$40 - but well worth it IMHO). See this sticky thread: http://forum.audacityteam.org/viewtopic.php?f=28&t=1994

This workflow tutorial that I wrote for the 1.3 manual may interest you: http://manual.audacityteam.org/man/Samp ... gitization

WC
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Re: Audacity appears to be increasing the bit rate?

Post by waxcylinder » Wed Jan 11, 2012 6:38 pm

This set of tutorials may be useful too: http://manual.audacityteam.org/man/Tuto ... iscs_to_CD

WC
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Re: Audacity appears to be increasing the bit rate?

Post by Submariner » Wed Jan 11, 2012 7:30 pm

Thanks "waxcylinder"

For the tips - appreciated. Like youI'd rather be listening than playing around with software.
But do you think it would sound better than the interim? DVD route I'm trying?

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Re: Audacity appears to be increasing the bit rate?

Post by waxcylinder » Wed Jan 11, 2012 8:14 pm

I think you hit the nail on the head yourself earlier in the thread - you want to avoid as many conversions as possible - so if your final target is 44.1 kHz CDs than it is better to stick with 44.1kHz throughout the cycle.

It is worth using Audacity at 32-bit float as I said earlier to provide max. headroom for processing and putting up with the one downsample to 16-bit for final export (and when I use ClickRepair I do so immediately after capture, I export a 32-bit float WAV to CR and import the repaired file also 32-bit float back into Audacity for all further processing).

I also forgot to mention that when I export to 16-bit WAV I have Audacity set to triangular dithering (accees control of this through Edit> Preferences )

WC
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Re: Audacity appears to be increasing the bit rate?

Post by Submariner » Thu Jan 12, 2012 6:33 pm

waxcylinder wrote:I think you hit the nail on the head yourself earlier in the thread - you want to avoid as many conversions as possible - so if your final target is 44.1 kHz CDs than it is better to stick with 44.1kHz throughout the cycle.

It is worth using Audacity at 32-bit float as I said earlier to provide max. headroom for processing and putting up with the one downsample to 16-bit for final export (and when I use ClickRepair I do so immediately after capture, I export a 32-bit float WAV to CR and import the repaired file also 32-bit float back into Audacity for all further processing).

I also forgot to mention that when I export to 16-bit WAV I have Audacity set to triangular dithering (accees control of this through Edit> Preferences )

WC
Once again thanks for the tips but what is " triangular dithering ".. I've never heard of that before.
Regards
Peter

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Re: Audacity appears to be increasing the bit rate?

Post by steve » Thu Jan 12, 2012 8:33 pm

"Dither" is a technique used to improve the conversion quality when down-sampling from a high bit format (bit depth) to a lower one.
http://en.wikipedia.org/wiki/Dither

In the case that waxcylinder is describing, the audio is being converted from 32 bit float to 16 bit.
16 bit audio is an "integer format" - that is, it uses whole number values between 0 and 65,535 (well actually +32,767 and -32,768), whereas 32-bit float can also represent fractional values which provides for much higher precision.

Clearly,when dropping down from 32-bit float to 16-bit integer there needs to be some method for "rounding" the sample values.
The most simple way to do this is to either round up, round down, or round to the nearest whole number.
As you can probably imagine, compared to the 32-bit waveform, the 16-bit waveform will be less smooth and there will be small "steps" appear where there were previously smooth curves. This is referred to as "quantize error" and it is a form of distortion that adds a little harshness to the sound.

It has been found that when rounding the 32-bit values down to 16 bits, this type of distortion can be eliminated by adding some randomness to the rounding. For example, when rounding 76.5, rather than always rounding to 77, it will sometimes round to 76 and sometimes round to 77.

The added randomness is called "dither" , and what it is doing is adding a tiny bit of "random" noise which helps to "smooth out" the steps between 16 bit values.

Even though it is helping to smooth out the steps and reduce the audible artefacts of quantize errors, it is obviously desirable to make the dither noise as unobtrusive as possible. To do this, the noise can be "shaped" so that frequencies in the noise are distributed into ranges where they are least noticeable. One of the most effective ways of doing this is with "triangle dither". Another good one is simply called "shaped dither". Both of these forms of dither produce good results.

Note that to minimise the dither noise, it should only be applied once, otherwise there can be a noticeable build-up of what should be insignificant noise.
If all of the processing is done at 32-bit and the audio is converted to 16-bit right at the end, then it will only be during that final conversion step that dither will be applied. The disadvantage of using 16 bit audio tracks in Audacity is processing is done at 32-bit, so dither will be applied with every processing step.
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