Best Sounding Software?

[The Computer Audiophile]

Hey Al - I really do think you're getting to where you want / need to be. It's pretty cool.

 

Bit perfect playback is a combination of hardware and software. It simply means that the digital stream coming from your computer is a bit for bit exact stream of the original source. Mac OS X outputs a bit perfect stream from the moment you turn it on. There are things you can do to screw this up, like adjust the iTunes volume and you do have to change the sample rate to match the content you are playing. If playing 24/96 music you need to tell the Mac that's what you're playing. With Windows you need to stream the digital signal "around" the kmixer or any resampler. Many people use ASIO, but not all sound cards or external DACs support ASIO. In that case you have software that is bit perfect but not the hardware.

 

Now go to bed! Just kidding of course :-)

 

[The Computer Audiophile]

Hey Paul - Now I see what you're getting at. Thanks for clarifying the STOCK v. STOCK thing with me. This should be a fairly easy test since no config has to be done. If we really want to limit possible hardware differences we can always do this on a Mac running Boot Camp. This way we have the same hardware but different operating systems.

 

Also, thanks for recognizing that it's much more laid back over here. I think we are all in this hobby for enjoyment of music and music reproduced at high quality. When it stops becoming fun and enjoyable to even talk about it with others who share the same likes and dislikes then we have a problem. We all have much better things to do than argue with each other over a single bit or byte.

 

Have a great holiday weekend at the cottage :-)

 

[JonathanCatuccio]

Hi guys - just to comment on ALAC playback in J. River:

 

you can use DSP-Worx package to get ALAC playback in J. River w/o Quicktime.

 

Since J. River lets you can configure DirectShow filters "in-house", we can select the DSP-Worx Bass_ALAC as the source filter.

 

Similarly, AAC (m4a & m4v) can be played back with Haali splitter (source filter) and FFDshow (audio decoder).

 

If Quicktime is installed, and the user delects m4a from DirectShow filter setting in J. River Media Center 12, MC will utilize the Quicktime engine.

 

I'm not 100% sure that we lose RG, visualation, etc. if MC needs to use Quicktime externally; I thought that was the case but a recent discussion called that into question.

 

Anyway, hope this helps.

 

DC

 

[The Computer Audiophile]

Very cool! Thanks for the information!

 

[Al]

So, if you turn the volume up in OSX...its not bit perfect anymore...? that would mean you have to have measure in place to make sure all tracks playback at the same volume souldnt it...?

 

it sounds attractive (as a long/medium term solution), but if it can be messed up easily, ill do it, so perhaps its not for me.

 

Anyway, i've led the thread way OT, I should have kept this in my other thread. Apologies to the OP and all!

 

[Al]

ont the previous page, re using winamp, that i used it to rip to flac yesterday, and noticed little or no difference between those rips and eac rips. however, eac has secure mode and pretty much promises perfect rips so, unless i discover winamp has a similar function (im still new to winamp and am learning somewhat haphazardly), will continue to use eac just for the added assurance.

 

though that might be useful to someone.

 

[The Computer Audiophile]

Hi Al - The volume in iTunes should be set at 100%. Then you just use the volume control on a preamp or DAC or even a pair of active speakers. It's not just iTunes that works this way. Software volume controls stink. But keep in mind that in many situations people do not have to have bit perfect output. Remember if it sounds good to you then it is good.

 

Playback at the same volume, also called normalizing, is the antithesis of good sound. This is another conversation altogether!

 

[Al]

Many thanks, i think i understand.

 

[JonathanCatuccio]

J. River's volume control is "32-bit and it works with line-out style cards that have no system volume. Some of the best soundcards have no system volume so that you get a bit-perfect signal with no system interference" ([email protected])

 

I've read discussions before on this topic, and I think it's rather strong to state it would be "the anti-thesis of good sound" to use a volume control, but I agree you want to avoid it if possible. Even 32bit processing will make fidelity take a hit at some point in software applied reduction/gain.

 

One of my MC zones is my crappy TV speakers via graphics card HDMI (wave) audio; I use internal volume control on this.

 

My ASIO output with the EMU1212M to Benchmark DAC1 is direct output to S/PDIF in EMU Patchmix; volume controls on my Sunfire Preamp are used here.

 

What I am doing, however, is after having calibrated the potentiometers on the DAC1 and measuring input voltage at my monoblocks, is use Replay Gain. I counter the reduction (which I think is excessive) with this technique, and it's pragmatic...I can't stand volume levels fluctuating all over the place.

 

I do cricitcal listening with this setup, but more often have music playing in the house while I'm working, so this just works for me.

 

We should discuss RG more though (i.e. is it as detrimental as internal volume controls?)

 

DC

 

[The Computer Audiophile]

Hi DC - Thanks for the post.

 

Maybe I should clarify my statement a little bit. In my opinion using volume normalization to make everything the same or similar volume is the antithesis of high quality sound. While normalization is very convenient for a ton of people it is almost the same thing as the Loudness Wars where engineers compress the heck out of popular albums to make everything sound loud. This you have normalized volume, but in reality nothing is loud and nothing is quiet when it's all the same level.

 

Controlling the volume through an application's volume controls is less of a problem even though it will result in a less than bit perfect signal.

 

I am a bit confused by Matt's statement. Maybe you can lend a hand to clarify for me :-)

 

"32-bit and it works with line-out style cards that have no system volume. Some of the best soundcards have no system volume so that you get a bit-perfect signal with no system interference" ([email protected])

 

When he says line-out style cards with no system volume control I think of every DAC I have connected to my music server. Each of these locks the volume. So, does he consider these as line-out "devices" with no system volume control? I'm really interested in this ans it would be pretty cool to get more info.

 

If I read the whole statement correctly it doesn't say anything more than the JRiver volume control is 32 bit. It does not say adjusting the volume through JRiver won't harm the sound. In a way it seems to say that you can use the J River volume when you use sound cards without system volume controls. What do you think, I'm interested in your opinion.

 

Don't get me wrong in this whole thing I think every feature has its place and if people enjoy the sound more then I am all for it!

 

Let me know what your take is on this one.

 

 

 

[JonathanCatuccio]

I think you've got it.

 

For example, my EMU when chosen as sytem default locks down (bypasses?) any Windows volume control. The volume bar in Conrtol Panel>Sound is greyed out. One needs to use Patchmix for any type of volume control, or use J. river internal volume control. System volume is not an option with EMU, but is with something like Realtek via MC volume control (user chooses between System or Internal, except on Vista there is now System, App, Internal IIRC).

 

My thought is that were the volume control 16-bit, you would see a degradation much quicker than when gain/reduction is applied at 32bit.

 

I remember reading about using Patchmix master fader to reduce or add gain, and someone had calculated (on Hydrogenaudio?) the bit loss as a reduction was applied with the master fader.

 

Similar reason (but others too) to record at high bitdepth and decent sampling rate (at least 96khz) before dithering down etc. to 16/44 if you choose to.

 

 

But regarding RG, it's not compression. It's not changing the waveform like a compressor does; and it's not really normalization because it's targetting -89db IIRC with a reduction of overal volume output. I don't think it's going in like a Normalizer and rasing the highest peak to zero etc.

 

best

DC

 

 

[JonathanCatuccio]

This might be helpful for the earlier discussion:

 

MS claims that Vista's resampling is of higher quality than XP's, but I have no personal experience of it.

 

On XP, if the sound card doesn't natively support the 44.1 kHz sample rate it may be better to use MC's software resampler at 48 kHz than let Kernel Mixer do that.

 

With a high quality sound card that supports 44.1 kHz also Direct Sound through XP's Kernel Mixer should be fine even though it may not be exactly bit perfect.

 

Usually ASIO is necessary only for bit perfect SPDIF pass through at 44.1 kHz. 48 kHz can be bit perfect without ASIO. At least that is the case with my Terratec DMX 6fire 24/96 sound card.

Alex B @ J. River Interact forums

http://yabb.jriver.com/interact/index.php?topic=43648.msg298290#msg298290

 

I'm also thinking that ReplayGain is more like the internal volume control of MC being applied to each track to hit the target output rather than normalization which would seek to make each track as loud as possible with clipping.

 

So, compression = can sound really good from what I've read but very invasive/destructive, normalization = maxmizing amplitude of one track to make peaks as close to 0db w/o clipping, and Replay Gain = high quality 32bit volume control to apply slight (-5 to -13db reduction) to hit a target volume.

 

Disclaimer, my comment about compression should not be read as support for the "loudness wars", rather that multi-band compression when used properly by good radio stations (or in MC with DirectX dsp plugin) might give new life to poorly recorded material. MusicHawk over at Interact (J. River forums) has a lot of experience with radio and has had songs by the Beatles, for example, "come alive". I'm not big on plugins and EQ at all , but I do have lots of 45s, LPs, and lossless files, on all of which can be some music that sounds just bad....because of poor recording techniques or equpiment? ...arguably because of compression used improperly in the first place? LOL. So I'm opened minded, like Chris, that sometimes post-processing can work, but one must be very prudent in its application.

 

 

Dc

 

[Paul Held]

Chris - Have a great 4th - we had ours on Tuesday (Canada Day)!

 

btw, new Athena floorstanders coming to boost the cottage system. Last month I got an Apple TV 40G unit for the cottage, being served off an old XP machine running iTunes and 320AAC files. I have it streaming from the XP machine and it works great.

 

What a fantastic piece of gear overall! The ATV is connected via HDMI to an older Sony LCD 42", running toslink into an Onkyo AVR. Sounds good, looks great!. The big Athenas will help make it all sound even better

 

talk to you soon

 

[rahan]

Hi,

 

I'm new to this forum too. I need a clarification wrt to what is the higher resolution formats that J River can recognize and play. I don't seem to be able to get j River to recognize or play the Reference Recordings HRx files in 24/176.4.

 

Is J River limited to 24/96 and lower? If there are any special settings required to get it to play 24/176.4, where can I get info on that?

 

The J River sounded great with all the formats I got below 24/96. I use the Stello DA100, DA 220 MkII for USB and the Weiss Minerva with Firewire. Both on Vista and DirectSound.

 

Thanks.

 

[lmpine]

Chris--

 

This is my first post over here, but I've been reading you site for a while and I've found it quite helpful. Keep up the good work!

 

I'm really excited about being able to use an IPhone to control ITunes and I was messing around last night AND stumbled upon something that I need some help with...

 

From all the research I've done (both on this site and elsewhere), I was completely convinced that I would not be able to get bit-perfect output on my current machine with Itunes (Windows XP without an Airport Express) which up until last night was running Foobar with ASIO. I am connecting to my DAC/processor with an Empirical Audio Off-Ramp Turbo (the original with M-Audio Drivers).

 

Last night I tried testing two .wav files encoded in DTS and DD played through ITunes 7.7 (with volume control set-to max). Much to my surprise, both files were played back perfectly by my processor which locked onto the signal and confirmed both bitstreams as DTS and DD.

 

Is this really bit-perfect and can I move forward with this set-up confidently without an Airport Express?

 

Thanks for any help or insight, Lewis

 

[The Computer Audiophile]

Hi Impine - Welcome to Computer Audiophile. As far as I know the only way to get bit perfect audio from iTunes on a PC is to use the Airport Express. The AE allows you to skip KMixer.

 

If you go forward without the AE you can't use iTunes if you want bit perfect audio :-( At least for now :-)

 

[illuminetics]

What a great site!

 

I prefer Winamp just because I've been using it for years and it supports my codec of choice: flac. Audio quality comes from the codec more than ther player software. I ripped all me CD's to flac files and it's open source - I love that. Suppose to be "lossless", but I think I can detect a very slight loss in audio quality while listening to the origional CD "red book" 16/44.1 vs. the compressed flac files. WAV files are uncompresssed and therefore truely lossless. Man: It's easy to get burried in this stuff.

 

BTW: Another major contributor to hi-fi quality comes from the ADAC process. I have an EMU 1820m and love it, but that's another discussion. So much to discuss!

 

Take care,

 

Randy

Bend, Oregon

 

[The Computer Audiophile]

Hi Randy - You're amongst us few audiophiles who dare say lossless compressed files may not sound as good as WAV of AIFF uncompressed files. Don't worry about speaking up around here though. I like to say we have the smartest forum readers around and they respect everyone's opinion whether they agree with it or not. This is a very laid back site and we don't like to take the fun out of an incredibly enjoyable hobby. Anyway, I have no doubt that FLAC and other lossless compression methods do store the file as a 100% perfect copy of the uncompressed version, but where the possible problems exist are in the decoding process on the fly during playback. I've talked to several manufacturers and even very high-end DAC designers who agree with this statement.

 

Thanks for the post.

 

[Innertuber]

Hi Randy

 

I like you use Winamp and FLAC on PC. It's easy and saves space. Once in a while I get a bad rip or screwed up folders, but I get that with iTunes too periodically.

 

[illuminetics]

Chris,

 

I got verbally abused at another site by suggesting that flac-encoded audio may be a little bit (no pun intended) lossy. I swear I can detect a slight degredatiopn of my flac audio vs. the origional CD played through the same DAC (and transport that encoded the CD to the drive).

 

I need to do some experiments. So much to learn... so little time...

 

Randy

 

[Innertuber]

I always thought the advantage of FLAC was smaller file with zero loss? Tell me that I'm not fooling meself!

 

[The Computer Audiophile]

Hi Innertuber - There are many things to this issue, only one of them being file size with zero loss. I think I am going to write an article on this one and discuss all the items involved in this. FLAC is certainly smaller and has zero loss as a stored file. It is a great transport file type because of these two things. Playback is another story and there seems to be arguments on all sides of this one. I chose to go with uncompressed AIFF because there is no argument against AIFF except file size.

 

Of course it all comes down to listener opinion. If FLAC is what people like than that's cool with me. Heck, if 128 kbps mp3 is what people like oh well. I'm sticking with AIFF :-)

 

Stay tuned for an in depth article.

 

[thermalbat]

I don't know if this has already been posted, but if not here's a really excellent

page on setting up EAC to rip a CD

 

http://www2.firehose.us:81/~jiggafellz/eac/index.html

 

[JonathanCatuccio]

Great tutorial, thanks.

 

I used to use EAC until I discovered J. River. Encodes to multiple formats, secure ripping, I can access YADB, FreeDB, and Gracenote for meta-data, and I can rip 4discs at once.

 

While they don't use some of the advanced settings like offset etc., I'm confident about the rips.

 

dc

 

[cowby]

Hi,

 

As a new member here. I'd like to share my experience on CAS (Computer As Source) with playback software. First of all, you should have a good DVD ROM/CDROM for ripping and I heard that Plextor and Yamaha has "audio" grade ROM drive which can help you ripping the song in a nice and clean way. But pls try your luck to find one cos' they are no longer avail. new in the market. Mine are LG and Pioneer - DVD Rom.

 

If for casual listening, I like to use Media Monkey or foobar2000 but recently, I started to use this to listen to my music (for serious listening). http://www.samplitude.com/eng/seq/

 

Mine version is a light version 8.x (which came with RME audio interface - Fireface 800) and I know this is a very expensive software (when FULL) to buy until you are a producer. See if you can buy a cheaper and lighter version.

I ripped the CDs to Wav format and import them to Samplitude. The nice thing is Sam can load your wav into RAM they called it RAM project. The benifit is you will get a very CLEAN sound from RAM rather than reading wav from a harddisk (mechanical movement or other annoying electronic hisss). And you know that nowadays 1G memory means nothing in terms of money. But the down size when playing song in SAM is you only get 2.xG (500MB for your own OS plus anti-virus and etc ..) of songs as Windows XP only support 3G of RAM (I have no idea on Vista) and it doesn't support playlist like Winamp or Foobar. So you need to drag your fav. songs in one by one within a single track. Then it will play the wav files in turn. You can also use this kind of Wav editor (e.g Sony's Sound forge) to trim the leading space or normalize the vol of the song. Fade in Fade out and make it like a "love song medley" then save it as a BIG wav file for future listen. Again with 2G free of memory, I think you can have around 3x songs to listen at any given time. Sounds not bad huh ?

 

P.S I am also hunting for a cheaper solution for any sound editor / music playback software which can do RAM streaming.

 

Regards!

Cowby