Question(s) about 24 bit and 16 bit

[2000_man]

I'm looking to burn a 24 bit show to DVD-Audio. I'm also wanting to convert the 24 bit to 16 bit so I can make a cd copy also. I read this thread

http://www.thetradersden.org/forums/showthread.php?t=39781&highlight=24-bit

about it also.

What I'm wanting to know is if anyone has used DVD-Audiofile and if it is a good program (or any others you have used from first hand knowledge). I was thinking about using it to burn the DVD-Audio. Also, what program would any of you suggest that would convert the 24 bit to 16 bit so that I can make a copy for a cd too. Thanks.

[Tubular]

I've used DVD-Audiofile (free) and it is great, it just takes 30-45 min. to create the iso file. You also need some extra space on your hard drive. If you are burning 4 GB of wav files, you will need an additional 4 GB for the DVD-Audio structure folder, then an additional 4 GB for the iso. The DVD-Audio structure folder gets erased after the iso has been fully created. Don't try stopping the process after the DVD-Audio structure has been created, and then try to burn that, it doesn't work. I tried burning this folder as data, and then as a UDF disc and my DVD-A player wouldn't recognize the disc. I think this is because a DVD-A disc needs some special identifying features that your typical burning programs can't add, but I'm not sure. However, if you let the authoring process fully complete, and then burn the iso with a burning program it will work flawlessly.

I think Audacity will convert 24 to 16 for free, and any program like Soundforge (not free) can also do it.

[2000_man]

Thanks Tubular. I figured you would know the answer. I've got plenty of storage so I luckily don't have to worry about that. I was going to use DVD-Audiofile and I have Roxio Easy Media Creator 9 to burn it. Does DVD-Audiofile create a true DVD-Audio or does it use DVD-Video to store the audio? I have a Panasonic DVD-Audio player that I'm going to play it on.

I know enough about the audio side to be fairly comfortable, but I'm still a novice at burning DVD's or DVD-Audio. I downloaded a show in 16 bit that someone converted from the 24 bit version and it had a lot of pops. So, I downloaded the 24 bit show which is flawless and I was going to experiment and burn my first DVD-Audio disc and then try and convert it to 16 bit for a cd. Thanks again.

[Tubular]

No problem. :D DVD-Audiofile creates a true DVD-Audio disc. All of the files are in the AUDIO_TS folder and not in the VIDEO_TS folder. They are also .aob files, not .vob files, like video would be. Remember to download and install the Java Runtime environment (free), as DVD-Audiofile needs this to work.

[2000_man]

Good deal. Got Java installed. I found the board where the developer talks about DVD Audiofile and I've read some on it. I downloaded Audacity last night and played around with it and got the converted the 24-bit to 16-bit. The newest beta version can convert flac files so I used that one and it worked great. That's a neat little tool. Thanks for the recommendation. Sounds flawless now. Whoever converted it the first time did something wrong because there were numerous pops that weren't on the original 24 bit. Now I can burn it to cd and dvd.

[direwolf-pgh]

:wave: question/help !!

Ive begun to create Audio DVD's with this program http://www.audio-dvd-creator.com.

Its great - I burned two 3disc shows & created a menu with the info.file on the DVD.
+ each show/track name shows original source

example of text on TV screen when playing:

Show: gd78-04-16.sbd.miller.xxxxx.sbeok.flac16
Track: gd78-04-16d1t11

Here Is The Question:

Check out Step 2. http://www.audio-dvd-creator.com/guide.htm
Audio Format:Select the audio encoding format: PCM (48 kHz/16 bit) has high quality which is similar to Audio CDs, the total playing time is up to 6 hours. PCM (96 kHz/24 bit) has the best quality, but total play time is only 2 hours
Yes I understand most everything I have is 44.1/16bit.

What I dont understand is:

Are there any cons to converting/recording the 44.1/16 wav files > 96/24 or 48/16 PCM

Seems to me it would be a bunch of extra 'headroom' not being utilized - but that is a swag. I played the disk on my best stereo system - sounded great :thumbsup (but that doesnt mean im not doing something 'wrong')

thanks.

[Tubular]

With DVD-Audio (not an audio only DVD-video) there is no need to upsample from 16/44.1 to 16/48. You can burn the CD quality 16/44.1 files right to DVD-Audio without upsampling.

http://www.hydrogenaudio.org/forums/index.php?showtopic=52933

"If you're playing back 16-bit 44.1kHz sampled content without any processing, and the DAC is ideal, then increasing the sample rate or bitdepth will make no difference to the output quality. If the DAC is non-ideal, then upsampling in software can improved the measured performance. If the DAC is terrible, then ABXing this imrpovement by listening is possible (especially with torture signals!)."

If you're going to upsample 16/44.1 to 16/48 to be able to make an audio only DVD-video disc, then I would use a software program like Audacity or Soundforge at the highest quality settings with anti alias filter. So you are saying that you can input 16/44.1 files into Audio DVD Creator and it will upsample to 16/48 for you, then author the DVD? I'm not sure how much of a quality difference there would be between Audio DVD Creator's upsampling and Audacity's or Soundforge.

If you upsample and bit expand from 16/44.1 to 24/96, then you will not be able to fit as many minutes of music on a disc, which is a major drawback, esp. if it doesn't make a huge difference in sound quality. Maybe you could try upsampling and bit expanding one 16/44.1 show that you have to 24/96 with Audacity or SF, then doing an A/B comparison. If the 24/96 sounds noticeably better with your DVD player & Digital to Analog converter combo, then go for it, if you don't mind fitting less minutes of music on a disc. My guess is it probably won't make a huge difference in sound quality even if you have a lower end player.

As for gear, I would get a DVD-A/V player or receiver with a real multibit DA converter, not a 1 bit converter with high sampling frequency. Denon, Onkyo, and Yamaha are good consumer brands. Parasound and Adcom are a couple of good affordable audiophile brands. www.stereophile.com and www.audiogon.com would definitely be worth checking out for used gear and gear recommendations. Careful, some DVD-A/V players will only play DVD-R and not DVD+R. www.ecost.com has good deals on new and refurbished gear. You may want to wait a couple years to buy a new standalone player until the format war that is currently going on between HD-DVD and Blu-Ray is over and a winner is declared. Then buy a player that supports DVD-Audio. It looks like the superior format Blu-Ray will be the winner as Blockbuster is backing it.

[Audioarchivist]

I may be way off base here, but wouldn't upsampling 16/44.1 recordings be pretty redundaant, the same as upsampling mp3 to wav is totally useless, or upsampling a video CD to a dvd video...
It seems to me that it would just makes those files bigger and bloated, I'm not too sure it would make the sound better, unless your ad/da converters are shite at 16/44.1 and awesome at 16/48.

[Tubular]

There are some DA converters that use interpolation:

http://en.wikipedia.org/wiki/Interpolation

"In the mathematical subfield of numerical analysis, interpolation is a method of constructing new data points from a discrete set of known data points."

So these interpolating, upsampling DA converters add the missing info, so they claim. Here is a good article on Simaudio Moon CD players, and why they use oversampling, not upsampling:

http://www.simaudio.com/upsampling.htm

"In fact, and most probably, these latest methods actually deteriorate sound quality if the conversion takes the sampling rate to a frequency that is not a direct integer multiple of the original sampling rate, being 44.1 kHz for audio CD."

A while ago I was wondering if converting from 44.1 to 48 would cause any quality loss, because this is done all the time for DVD-videos. Someone films a show and then they need taper's rig audio to synch to it, cause camera mic audio sucks. So they find a 48 > 44.1 FLAC or SHN set, then upsample to 48. I did some listening tests (on PC speakers) between:

1)44.1 FLAC set

2)44.1 FLAC set > upsample to 48 in Soundforge at highest quality settings w/anti-alias filter.

#2 sounded duller in comparison to #1. I liked #1 better.

[Tubular]

Holy crap, I just upsampled a 16/44.1 wav to 16/48 with Wavelab and played it back in Winamp and the difference is incredible!!! :wtf: The 16/48 sounds unbelievable compared to the 16/44.1! I definitely prefer the upsampled wav! I guess it is because I'm using the stock crappy soundcard that came in my Dell laptop! The Hydrogen forum quote is right!!! If you have a crappy soundcard or DAC, upsampling will definitely improve the sound quality. I know it isn't a 24 bit soundcard because it won't play a 24 bit wav in Windows Media Player. The PC I did the other test on did have a 24 bit soundcard, and I guess it was a much better soundcard with a much better DAC, so I guess I noticed the non-integer multiplication of going from 44.1 to 48.

This is big news IMO for anyone with a crappy DAC! Try upsampling a 16/44.1 wav to 16/48 on your PC and see if it sounds better! Holy shit! :lol :lol

[direwolf-pgh]

So you are saying that you can input 16/44.1 files into Audio DVD Creator and it will upsample to 16/48 for you, then author the DVD? I'm not sure how much of a quality difference there would be between Audio DVD Creator's upsampling and Audacity's or Soundforge. Yes. You collect the wav files & the program will upsample to PCM, then author the DVD.
Im just not familiar what 'method' or codec they are using. I was impressed with the results.
If you upsample and bit expand from 16/44.1 to 24/96, then you will not be able to fit as many minutes of music on a disc, which is a major drawback, esp. if it doesn't make a huge difference in sound quality. Maybe you could try upsampling and bit expanding one 16/44.1 show that you have to 24/96 with Audacity or SF, then doing an A/B comparison. If the 24/96 sounds noticeably better with your DVD player & Digital to Analog converter combo, then go for it, if you don't mind fitting less minutes of music on a disc. My guess is it probably won't make a huge difference in sound quality even if you have a lower end player. :lol: Im not a broke college student. Im playing back on a mid-range $5k component stereo system. The dvd-a/sacd/dts is a newer sony...nothing newsworthy but not a pos either. Most of my concern is about the upsample conversion - who has the best codec - and/or is this procedure technically/sonically 'wrong'.

The only reason Im interested in this procedure is having one concert, seamless on one disc. (sometimes having second set on 2discs is a bummer)
I may be way off base here, but wouldn't upsampling 16/44.1 recordings be pretty redundaant, the same as upsampling mp3 to wav is totally useless, or upsampling a video CD to a dvd video...
It seems to me that it would just makes those files bigger and bloated, I'm not too sure it would make the sound better, unless your ad/da converters are shite at 16/44.1 and awesome at 16/48. This is what I believed - just want to ensure Im not degrading the 'soon-to-be considered lossy' 44.1/16 stuff any worse than it is.

Holy crap, I just upsampled a 16/44.1 wav to 16/48 with Wavelab and played it back in Winamp and the difference is incredible!!! :wtf: The 16/48 sounds unbelievable compared to the 16/44.1! I definitely prefer the upsampled wav! I guess it is because I'm using the stock crappy soundcard that came in my Dell laptop! :wtf: you cant sample music on laptop speakers - you are fired! ;) :D

[Tubular]

Sony doesn't make DVD-A/V players, unless they just started making them. SACD was/is a competitor to DVD-A, so Sony was pushing SACD exclusively. If your player is a DVD-V/SACD/CD player, then it is likely to have a 1 bit/high sampling rate Digital to Analog Converter only, because SACD is a 1 bit system. A lot of conventional CD players also use a 1 bit sigma delta oversampling DA converter as well, even though CDs are 16 bit. If you are running analog outputs from your disc player to your receiver, then upsampling 16/44.1 > 16/48 will most likely improve the sound quality, because, no offesnse, your DA converter is less than ideal. I spent a lot of money on an SACD player and about 20 SACDs myself. :disbelief A multibit sigma-delta oversampling DA converter is much better and is found in higher end CD players and most DVD-A/V players (I think, at least the higher end ones). Companies use 1 bit converters because they are cheaper than multibit ones.

If you are sending a digital signal from your DVD player to your receiver or preamp/processor, then it depends what kind of DA converter is in that component.

It would be a good idea to burn a CD of a show and then burn an audio only 16/48 DVD-video with Audio DVD creator. Then do a listening test and decide what is best for you. Here is the thing, though: If you ever upgrade to a Blu-Ray/DVD-Audio/Video player with a true multibit converter, then those upsampled 16/48 audio only DVD-Vs will sound WORSE than CDs because you now have a great DA converter. Upsampled files will no longer sound better. In fact, upsampled files will sound worse unless they are a direct integer multiple (2x, 3x, 4x, etc.) of the original file:

"When changing the sampling rate, it is better to maintain an integer multiple of the original signal's sample rate, so the processing is kept simple. More importantly, the end result is more accurate, thus enabling a higher fidelity of sound reproduction. A two times (2x) oversampling system will double the sampling rate, by adding one easy to find numerical value in between each actual sample. For example, when a 44.1kHz digital signal is processed, a 88.2kHz digital signal is obtained. It is simple, effective and precise because it is a direct multiple of the original digital signal. For an upsampler to make a 96kHz digital signal from a 44.1kHz signal, it will have to perform awkward mathematical operations to obtain a 96kHz signal. (96kHz / 44.1KHz equals 2.1768707…). This results in a less accurate output from the digital filter, with everything else following (i.e. digital-to-analog conversion and analog filtering) also being less accurate. As well, exactly like oversampling, the artificially higher sampling frequency created by an upsampler doesn't increase the actual frequency response of the system, but simply increases the lower limit of the frequencies that need to be eliminated."

So it would be a waste of a lot of discs (not THAT much money, I know, but it adds up after a while) to convert 16/44.1 to 16/48 if you ever get a Blu-Ray/DVD-A/V player with a multibit converter, or a receiver or pre/pro with multibit converters. You can burn a DVD-A disc and keep the files at 16/44.1.

I read that there is a great free program to resample, it is supposed to be better than Audacity, I'll look around for it. Wavelab is better than Soundforge from what I've read. I think foobar2000 will perform upsamlping to 48kHz on the fly, which improves the sound on my crappy soundcard/DA converter when listening through headphones. I'll have to do some listening tests and see how foobar2000 compares to Wavelab's upsampling.

[direwolf-pgh]

Sony doesn't make DVD-A/V players, unless they just started making them. SACD was/is a competitor to DVD-A, so Sony was pushing SACD exclusively. If your player is a DVD-V/SACD/CD player, then it is likely to have a 1 bit/high sampling rate Digital to Analog Converter only, because SACD is a 1 bit system. A lot of conventional CD players also use a 1 bit sigma delta oversampling DA converter as well, even though CDs are 16 bit. If you are running analog outputs from your disc player to your receiver, then upsampling 16/44.1 > 16/48 will most likely improve the sound quality, because, no offesnse, your DA converter is less than ideal. I spent a lot of money on an SACD player and about 20 SACDs myself. :disbelief A multibit sigma-delta oversampling DA converter is much better and is found in higher end CD players and most DVD-A/V players (I think, at least the higher end ones). Companies use 1 bit converters because they are cheaper than multibit ones.

If you are sending a digital signal from your DVD player to your receiver or preamp/processor, then it depends what kind of DA converter is in that component.

It would be a good idea to burn a CD of a show and then burn an audio only 16/48 DVD-video with Audio DVD creator. Then do a listening test and decide what is best for you. Here is the thing, though: If you ever upgrade to a Blu-Ray/DVD-Audio/Video player with a true multibit converter, then those upsampled 16/48 audio only DVD-Vs will sound WORSE than CDs because you now have a great DA converter. Upsampled files will no longer sound better. In fact, upsampled files will sound worse unless they are a direct integer multiple (2x, 3x, 4x, etc.) of the original file:

"When changing the sampling rate, it is better to maintain an integer multiple of the original signal's sample rate, so the processing is kept simple. More importantly, the end result is more accurate, thus enabling a higher fidelity of sound reproduction. A two times (2x) oversampling system will double the sampling rate, by adding one easy to find numerical value in between each actual sample. For example, when a 44.1kHz digital signal is processed, a 88.2kHz digital signal is obtained. It is simple, effective and precise because it is a direct multiple of the original digital signal. For an upsampler to make a 96kHz digital signal from a 44.1kHz signal, it will have to perform awkward mathematical operations to obtain a 96kHz signal. (96kHz / 44.1KHz equals 2.1768707…). This results in a less accurate output from the digital filter, with everything else following (i.e. digital-to-analog conversion and analog filtering) also being less accurate. As well, exactly like oversampling, the artificially higher sampling frequency created by an upsampler doesn't increase the actual frequency response of the system, but simply increases the lower limit of the frequencies that need to be eliminated."

So it would be a waste of a lot of discs (not THAT much money, I know, but it adds up after a while) to convert 16/44.1 to 16/48 if you ever get a Blu-Ray/DVD-A/V player with a multibit converter, or a receiver or pre/pro with multibit converters. You can burn a DVD-A disc and keep the files at 16/44.1.

I read that there is a great free program to resample, it is supposed to be better than Audacity, I'll look around for it. Wavelab is better than Soundforge from what I've read. I think foobar2000 will perform upsamlping to 48kHz on the fly, which improves the sound on my crappy soundcard/DA converter when listening through headphones. I'll have to do some listening tests and see how foobar2000 compares to Wavelab's upsampling.
:popcorn: sounds good to me.. but a few questions please.

1. DVD-A = backward compatibility with almost any DVD player.
The introduction of the DVD-Audio format required some kind of backward compatibility with existing DVD-Video players. To address this, most DVD-Audio discs contain, at a minimum, a Dolby Digital 5.1-channel audio track on the disc[2] (which can be downmixed to two channels for listeners with no surround sound setup). Some discs also include a native Dolby Digital 2.0 stereo, and even a DTS 96/24 5.1-channel, audio track[3]. I have a few and they play fine on a Sony deck - usualy its the DTS track from the disk.

2. i doubt I could hear the difference between 44.1/16 & 48/16.

3. what is your background in audio? I enjoy your comments.
Personally Im just a hobbyist that reads here and there & likes to fiddle with audio recording programs & gear.

4. Technology starts>changes>and usually ends up full circle.
Its hard to say what 'works' and what doesnt. I remember my first CD player in 1980's had a 20bit - now today its 1bit - tomorrow its whatever.
this is a random interesting read http://www.enjoythemusic.com/magazine/equipment/0506/zero_oversampling_dac.htm

appreciate the good thoughts :thumbsup

[Tubular]

This is a very good free sample rate converter, better than Audacity:
http://www.voxengo.com/product/r8brain/
I bet the pro version is even better, but it is not free.

1.You should hear those commercial DVD-As in all their lossless MLP or uncompressed PCM glory on a DVD-A player! "Fragile" by Yes, "Brain Salad Surgery" by ELP, and "Gaucho" by Steely Dan are just amazing. IMO, in my listening experiences, SACD=shite. It is a little bit better than CD, but not the huge improvement that was expected. SACD even has less resolution than CD above 10kHz. DVD-A on the other hand is a giant leap beyond CD and SACD resolution. You can read more about SACD vs. DVD-A vs. CD here: http://sound.westhost.com/cd-sacd-dvda.htm
The author really blows the lid off the scam that is SACD.

2.Upsampling from 44.1 > 48 degrades sound quality (unless you have a cheap soundcard, then it actually improves it! I don't understand this). In one listening test I did on a PC with a good soundcard it was clear that the upsampled wav sounded a bit dull compared to the 44.1 wav. Most DAT recordings originated at 48. Then to burn to CD, they are downsampled to 44.1. Now that DVD-video is here, they are going back to 48. It would be great if everything that was recorded at 48 could be circulated that way, but it probably won't ever happen for at least half of the seeds.

3.I'm just a hobbyist too. My interest in sound really took off after I bought a pair of really nice 4 way speakers about 10 years ago. They were very revealing, and I realized that my system sounded like shit!!!.....unless I played vinyl. Vinyl/analog was the answer! It didn't help that I had one of the harshest sounding CD players ever made, I think, a 1994 Sony. I hooked it up again a couple of years ago and it was painful to listen to. Now with DVD-A I'm very happy with digital.

4.I agree, we started with analog > digital > more lifelike digital intended to sound more analog. The r8brain sample rate converter page says it supports bit depths of up to 64! I hope I get to hear 64 bit recordings in my lifetime. I kinda doubt it though. If you have heard an analog sourced vinyl record, though, you have heard the best sounding recording that will ever exist, unless you are listening to master 2" reels. I read about half of that article, I'll read more tomorrow. I've never heard of the R2R zero-oversampling DA converters! Sounds interesting. :thumbsup

[Tubular]

Ooh, just found out about a big drawback to upsampling, at least with some shows. If your show has some tape hiss, then that hiss will be magnified significantly when you upsample. I just upsampled a track with some tape hiss from 44.1 > 48 with Wavelab and while it did sound better on my crappy PC soundcard/DAC, hiss increased a lot.

[Tubular]

Some additional thoughts on the Dolby Digital and DTS tracks on DVD-Audio/Video discs:

http://www.spannerworks.net/reference/10_1a.asp

"Both Dolby Digital and DTS are capable of 24-bit resolution, but currently nominally operate at 18-bit resolution, allowing a dynamic range of approximately 108dB. Theoretically, 24-bit resolution allows dynamic range of 144dB which, though higher, would be indistinguishable from the lower 108dB figure given the current limitations of playback hardware. For all practical purposes, both Dolby Digital and DTS Digital Surround operate at near, or above, 18-bit resolution and dynamic range (108dB)."

So in addition to being lossy, DTS and Dolby Digital (DTS less lossy than Dolby Digital) are only about 18bit/48kHz. Most DVD-Audio discs are 24bit/48kHz or 24bit/96kHz, and lossless. Dynamic range aside, you'll hear a world of difference between 18bit and 24bit, because the waveforms are more accurately described at higher bit depths.

http://www.dtsonline.com/consumer/technology/at-a-glance.php

The DTS 96/24 codec (lossy) is extremely rare and only found on a few receivers and players. It differs from the much more common regular DTS codec described above. You need a special receiver or player that decodes DTS 96/24. It is backwards compatible with regular DTS, but it will be downsampled to 24bit/48kHz. I haven't seen any discs (audio or video) that utilize this codec. Doesn't mean they don't exist, though.

The new codec, DTS HD Master Lossless (24/96 x 5.1 or 7.1? channels) is used on some HD-DVDs and Blu-Rays.

[direwolf-pgh]

yeah.. but the video industry needed that to fit the 5.1 streams due to DVD space limitation of the time.

Im kinda excited for 96/24 DVD-A. but Im not looking forward to the $$$ again for the 4th version of the White Album :lol: I guess its a good thing..an so the wheel rolls on.

-what are you playing your DVD-A on.. cause I know that laptop wont do it.

[Tubular]

My DVD-A/V player is a Denon DVM-2815. I got it here: http://www.ecost.com/ecost/search/search.asp?NavID_Search=false&CurDSN=simple&calledfrom=1&incimage=on&Search=Denon+DVM-2815&submit1=find a couple years ago for like $200 I think. Now it is $170. It is great, it plays DVD+R and DVD-R, Faroudja progressive scan, 5-disc changer, Burr Brown 24/192 DAC, built in Dolby Digital and DTS decoding, plays mp3s and wma8, coaxial and optical output. It doesn't have any of the new upconverting video equipment that newer DVD players have, but I don't have an HDTV, so it doesn't matter to me. It won't play region free PAL discs though. Not sure if that method to trick a PAL disc to play on an NTSC player works on it.

To buy a new Denon DVD-A/V/SACD player that plays DVD+R as well as DVD-R you have to spend a little more, like $300:
http://www.ecost.com/ecost/search/search.asp?NavID_Search=false&CurDSN=simple&calledfrom=1&incimage=on&Search=Denon+DVD-2910S&submit1=find
It's got all that upconverting gear and HDMI. It's got HDCD decoding too (mine doesn't).

When I get an HDTV in a few years, I'm going to get a Blu-Ray/DVD-A/DVD-V player with the best HDMI output standard that is capable of outputting the high resolution lossless Dolby and DTS formats. Also the player should have internal decoding of the high rez lossless Dolby and DTS formats, so you can use the 5.1 analog RCA outputs. This way I'll be future proof and I won't have to buy a new receiver right away. I don't think any of the current Blu-Ray players will do this.

The public didn't embrace DVD-A and SACD, but they will surely embrace Blu-Ray. So they're gettin' high resolution lossless digital audio whether they know it or not. :lol

[Tubular]

This is a good site to check out all the new Denons:

http://usa.denon.com/ProductDetails/DVDPlayersAndChangers.asp

If cost is of little concern there is the DVD-5910CI. Only $3800 MSRP :lol But I'm sure there are Audiophile players that cost 20 or 30 grand. :wtf: :lol

[2000_man]

Hey Tubular, got another question. I converted my 24-bit show to 16-bit and then corrected the sector boundary errors in TLH. Then I decoded them to wav files using Flac frontend. Then I burned the cd's. However, I'm still getting some very minute pops between tracks like I have sector boundary errors. I'm not getting them on all of the tracks, but quite a few. The original 24-bit flac files had no sector boundary errors. After I converted them from 24-bit to 16-bit in Audacity, they all had sector boundary errors so I ran all the tracks through TLH and all of them got fixed (supposedly). The show still sounds pretty good and it's better than the original 16-bit someone else put up but I'm a perfectionist and want the pops gone.

By the way, when you get your HD set, I highly recommend the Sony SXRD. I got the 60" rear projection last year and I love it. I wonder why I waited that long to get one.

[Tubular]

Did you run a len check with shntool on the files to see if they were SBE free? If you fixed the SBEs with TLH then I'm not sure what is causing the pops. :hmm: If the pops are only occuring at the transition from one track to another then that sounds like SBEs. Are you sure there are no pops on the 24 bit files? Might be mic overload or something else that happened during the recording of the show.

[2000_man]

Yes, I ran the len check with TLH. I did it after I converted the 24-bit to 16-bit and it had sector boundary errors. I fixed them with TLH and was error free. Then when I burned the cd's they came out with pops in between the tracks on some of the tracks. Someone who had uploaded the 16 bit version of the show had numerous pops within the songs so I decided to download the original 24-bit version to see if it was in the original recording. It wasn't so I converted to the 16-bit. Mine are only occuring between tracks which sounds like SBE's, but the len check says everything is fine. That is what has me confused.

I haven't burned the show to DVD from the 24-bit so I guess they could be on there too. The len check on the 24-bit version was error free.

[Tubular]

I would test the files again after they are fixed by TLH and make sure there are no "b's" in the left hand column of the len check. What burning program are you using? Maybe you burned a CD at too great a speed? You could try burning at a lower speed, like 8X. Also make sure the program isn't addding 2 second gaps between tracks. EAC is a good free burning/extracting program (make sure to uncheck the "add 2 second gaps on append" in layout): www.exactaudiocopy.de

[2000_man]

I tested them after I converted them and then ran them through TLH and fixed the SBE's with a len check and it said there were no errors. I use Roxio Easy Media Creator V9 and have never had a problem before. I burned at a high rate and a low rate and get the same problems. I downloaded EAC and even burned the first disc and I'm still getting the same problems. There seems to be a glitch in the transition of the tracks. Once the track is changed from one to the other there seems to be a millisecond of silence where the track would normally skip. The 2 second gaps were removed on both Roxio and EAC.

I'm beginning to wonder if the problem was with how Audacity converted the 24-bit to 16-bit. I've never had a problem with an original 16-bit recording before. I've burned many a show with Roxio and never had the errors I'm getting with this one which leads me to believe that the conversion didn't work as I thought. It converted the tracks, but there are errors that seem inherent in the tracks themselves.

[Tubular]

You could try burning a DVD-A disc of the show with DVD-Audiofile and see if there are gaps on the original 24 bit files. DVD-Audiofile doesn't add gaps between tracks. Also you could look at the files with CD-Wav editor (free). You can look at the wavs and see if there is any silence at the beginning or end of a file.

I've never used Audacity for converting from 24 > 16 bit. I have only used Wavelab, and I've never burned the files to audio CD. It could be you had some settings wrong in Audacity? Also you could decode the 16 bit files to wav and see if there are any other problems with the len check, like non-canonical header errors. These will show up in the middle column of the len check as an "h"

Just thought of this: you said you used a FLAC plugin for Audacity to do the conversion from 24 > 16. That plugin could be the problem right there. I would decode the original 24 bit FLACs to wav, then load the wavs into Audacity, then dither and convert to 16 bit wav. Then load the wavs into Trader's Little Helper to fix any errors. Finally, burn to audio CD. Hope that helps. :thumbsup

[Tubular]

So you don't have to waste another CD: use shntool join mode to merge the 16 bit wavs together into one big wav. Then listen to the wav with Windows Media Player or Winamp, or Foobar2000, and listen for any gaps between songs. Or look for gaps with CD-Wav editor.

[2000_man]

O.K. here is what I did. I converted the 24-bit flac files to wav files and then ran a len check. Everything was o.k. No errors. Then I took each 24-bit wav file and converted them to a 16-bit, 44 kHz sample rate wav file in Audacity. I then ran another len check and everything was o.k with no errors. However, when I went to listen to each wav file there now is a 2-3 second gap at the end of each wav file. I don't know enough about Audacity to know if there is a setting that I should've ticked off or on so I wouldn't get the 2-3 seconds of silence at the end of each wav file. I tried burning with EAC and the gaps are still there at the end of each song (like I figured since what you hear on a wav file is what you'll get. I guess I was hoping EAC would do something for me.). I've started a nice collection of cd coasters. Thankfully, they're cheap. :lol:

I simply imported each file and then set the project rate at 44. I then exported the file as a 16-bit wav file. Nothing more than that. I'm sure there is something else that needs to be done but I'm an Audacity neophyte.

I checked the preferences to see if there was any setting and I don't see one. The default under latency for audio to buffer is 100 milliseconds. Latency correction is set at 0 seconds. There's a heading called "Quality" under preferences and there's a conversion section. Those settings are set at "Real Time Sample Rate Converter = Fast Sinc Interpolation" and the Dither is set to "None". The "High Quality Sample Rate Converter = High Quality Sinc Interpolation" and the Dither is set to "Shaped". Whatever the hell that means :wtf:

I'm sure it's something I'm not doing (or should be doing) in Audacity. I just wish I knew WTF it was!

[Tubular]

Yeah, you want to use the high quality sample rate conversion and also use dithering. Dither adds a bit of low level noise (inaudible, I think) and makes converting from 24 > 16 more accurate.

You should send a private message to Five and AAR.oner (both moderators, you can find their names at the bottom right hand corner of the main technobabble page, then click on them and send a PM). They have used Audacity extensively and should be able to help you out. Then post back here with the solution. :thumbsup

[diggrd]

Just a quick question, when you look at the 44.1/16 wave in Audacity do you see 2 seconds of silence at the end of each file, or does the waveform look normal to the very end?

[2000_man]

I was hoping I could figure it out or someone could lead me in the right direction on here before I had to PM someone.

Just a quick question, when you look at the 44.1/16 wave in Audacity do you see 2 seconds of silence at the end of each file, or does the waveform look normal to the very end?

I imported the files and you can see the silence at the end of each wav. Can I (easily) edit that out and fix the SBE's once I do that?

[Tubular]

You could send them a PM, I'm sure they would be more than happy to help you! :) I'll download Audacity later and see if I can help you.

[Tubular]

In edit > preferences > quality > default sample format > leave at 32 bit float.

In edit > preferences > quality > default sample rate > leave at 44.1 kHz

In edit > preferences > file formats > uncompressed export format > leave at 16 bit

with the wav file in the window, on the file pulldown menu where you see the downward facing triangle > set sample format > leave at 32 bit float.

with the wav file in the window, on the file pulldown menu where you see the downward facing triangle > set rate > 44.1

In the bottom left hand corner, where it says "project rate", set at 44.1

Then go to file > export as wav

I just converted one single 24/48 wav file to 16/44.1, and there was no silence at the beginning or end of the file, and it sounds great. I used the high quality sample rate settings and triangle dither for both real time and high quality. I think 32 bit float precision is just the internal processing setting of Audacity. The higher the depth, the more precise it is. Hope this helps. :cool: :D

[2000_man]

What version of Audacity are you using Tubular? I had downloaded the 1.3.3-beta version because it could handle flac files. I'm going to uninstall it and go with the last workable version 1.2.6 and see if that works.

[2000_man]

I believe I have found the culprit. There is a bug in Audacity as explained in this thread (http://audacityteam.org/wiki/index.php?title=Splitting_recordings_into_separate_tracks) . Read under "Extra notes about burning to CDs" specifically #2 where it states:

2- Make sure your Project Rate (Hz), as shown in the Project Rate button bottom left of the Audacity screen, matches the rate showing in the Track Panel (where the mute/solo buttons are). If you don't do this, and you're exporting WAV or AIFF files, silences may be added at the end of tracks and the labels may not export in the correct position. This is due to a bug when resampling is done between the Project and Track Rate upon export. If necessary you can select your track by clicking in the Track Panel then Project > Quick Mix, which will resample the track to the Project Rate. Then simply delete the excess silence generated at the end of the track and the export will have no added silence. In 1.3.3, click Tracks > Resample, and in the box that pops up, enter the rate showing in the Project Rate button.

My problem is that if I do as you were saying in "with the wav file in the window, on the file pulldown menu where you see the downward facing triangle > set rate > 44.1"

then the track sounds slow as molasses. I have to only change the project rate to 44100 and I keep the set rate -> 96000 in the downward facing triangle. Since they both need to be the same because of the bug that will result in silence, it won't work. So, I have to delete the excess silence in which they make it sound easy, but I have to magnify the track to the nth degree to see the last bit of sound to trim the silence off.

[Tubular]

:lol Slow as molasses is right! I just tried a 24/96 wav and got the same results as you. :wtf: I guess I didn't notice the slowdown of going from 48 > 44.1 when I did it earlier, but I did notice it when I did 48 > 44.1 again. Also 48 > 44.1 slows it down less than going 96 >44.1. There must be some setting that we are missing. Either that or Audacity is a piece of shit for resampling!! You shouldn't have to edit out the silence at the end of a track for every wav!! :down: :rolleyes: Not to worry, there is a much better free sample rate converter here:

http://www.voxengo.com/downloads/?highlight=r8brain

Get the Win GUI version. Use the very high quality setting for best results. It takes quite a while to convert, but there is a batch mode, so you can input all your files and start it, then come back in like an hour and it will be done. You can still use Audacity to dither and convert 24/96 > 16/96. Just keep the sample rate the same. I'm not sure what the best dither setting is: triangle, rectangle, or shaped. r8brain will convert bit depths for you but I don't think it dithers, so it would be good to keep Audacity for dithering. I took a 24/96 wav, then dithered and converted to 16/96 with Audacity, then resampled to 16/44.1 with r8brain and the results are great. No slowdown, and no silence was added at the beginning or end of the track. :thumbsup

[Tubular]

Correction, r8brain also dithers:
"Fixed bug in implementation of dithering. Now dithering also incorporates a slight noise-shaping."

http://www.voxengo.com/product/r8brain/

So if all you need to do is to convert 24/48 and 24/96 files to CD-quality 16/44.1, all you need is r8brain! :clap:

[2000_man]

r8brain looks like the holy grail to a novice like myself. Thanks! :clap: I just wish I had found it about a week ago :lol

I found a really good forum over here (http://taperssection.com/index.php/topic,71191.0.html) when I was searching on what to do in Audacity. It covers just about every scenario possible. Sounds like triangle dithering was the way to go. I thing shaped is the default. When editing my wav files, I dithered with shape and then with triangle when I took out the excess silence. According to the tutorial, I should've gone triangle and then none. Anyway, I made a copy of the 16 bit converted from the 24 bit that I did in Audacity and I now have no pops, no silence and everything changes seamlessly. So, now I know what to do (and not to do) in Audacity and r8brain. I'm going to make a copy using r8brain and compare the two.

I learned some new things playing around with Audacity though that I didn't know before. Searching and reading the forums over there, r8brain seems to be pretty well accepted among the masses if all you are doing is a simple conversion. Audacity is more robust and you can play around with the dithering, but for my needs r8brain should do the trick from now on. And you don't have to edit out the silence at the end. ;)