The sane amongst us adopt the horses for courses approach. On a PC one uses a fomat native to the PC. With outboard EQ one would use a format native to outboard EQ. Only a complete eejit would seek to transfer MP3 as blocks to an outboard EQ. If an outboard EQ is processing audio the odds are the data arrived via one of the serial digital audio formats.
Given a level playing field i.e. dedicated hardware decompression engine connected to a source of MP3 data, dedicated hardware will run rings round a PC.
Given a level playing field i.e. dedicated hardware decompression engine connected to a source of MP3 data, dedicated hardware will run rings round a PC.
Regarding PC digital equalisation & room correction, check this thread at avsforum:
Digital Room Correction
Detailed step by step guide
Regarding Ogg vs WMA (lossless) formats, not much difference between the two as they are both lossless, however I have faith that the WMA format will be around for a long time.
Regards,
Dean
Digital Room Correction
Detailed step by step guide
Regarding Ogg vs WMA (lossless) formats, not much difference between the two as they are both lossless, however I have faith that the WMA format will be around for a long time.
Regards,
Dean
deandob said:
I've been through it (the first two pages of the thread and also the step by step guide), and it seems that the method doesn't take in mind the non-linearity of the soundcard. If the used soundcard is linear, this shouldn't form a problem.
But moreover: (correct me if I'm wrong) this method tries to flatten the sound 100%. This will not sound pleasant at all. You'd want to add a target curve after flattening.
See: http://www.prijsindex.net/tmp/room acoustics and eq.html
Because of this (and also to be able to experiment, especially in the bass region, which is a majorly tricky area, and cannot be dealt with measure-wise) I don't fancy the method.
What I want is a 32-band graphical equalizer without delay.
(I've tested Shibatch Super EQ plugin for Winamp 2 before, and although it's practical in usage, it's not HI-FI material: the sounds loses detail (~unlike my Behringer 8024)).
I'd really like to find one for PC, so I can also start to equalize at systems besides my own.
The delay-thing could be worked around (introduce delay for video also), but the loss of detail is an absolute no-go.
I have little faith in Microsoft on being a reliable party. Also: I really, really don't like Windows Media Player, and have therefore never liked the WMA-format. Maybe it's because wma-files are created using this software (right?), and I can't take those people serious (can I?).
Cheaper and simpler?
I also find this a very iteresting thread. I have been reading for a while some threads here and on the avs forum and have decided to give it a try.
A fully implemented computer stereo-system (as i'd like it
) could do digital crossover (2 or 3 way) & dsp (both available now with BruteFir on Linux, or only dsp with DRC on Win) to feed an external multi-channel non-os DAC (tda1541 for mid-highs, tda1543 lows). Now, the link between sonundcard and DAC could be spdif, USB or even i2s. It was suggested on this board that an expensive pro-audio card could be spared and get the I2s out of any non-resampling Envy24 based card, such as the Chaintec AV-710 ($~21) or Terratec Aureon (or on the CMI8738). That would be the cheapest and probably simplest implementation.
Anyone done something similar (i2s)?
these two theads are usefull:
http://www.diyaudio.com/forums/showthread/t-22019.html
http://www.diyaudio.com/forums/showthread/t-22556.html
I'll be doing some reaserch before I start ordering what i need.
Regards,
I also find this a very iteresting thread. I have been reading for a while some threads here and on the avs forum and have decided to give it a try.
A fully implemented computer stereo-system (as i'd like it
) could do digital crossover (2 or 3 way) & dsp (both available now with BruteFir on Linux, or only dsp with DRC on Win) to feed an external multi-channel non-os DAC (tda1541 for mid-highs, tda1543 lows). Now, the link between sonundcard and DAC could be spdif, USB or even i2s. It was suggested on this board that an expensive pro-audio card could be spared and get the I2s out of any non-resampling Envy24 based card, such as the Chaintec AV-710 ($~21) or Terratec Aureon (or on the CMI8738). That would be the cheapest and probably simplest implementation.Anyone done something similar (i2s)?
these two theads are usefull:
http://www.diyaudio.com/forums/showthread/t-22019.html
http://www.diyaudio.com/forums/showthread/t-22556.html
I'll be doing some reaserch before I start ordering what i need.
Regards,
swak said:Nielsio:
DRC allows any target curve characteristics, it even has a built in euphonic curve - according , to psychoacoutical tests, that is
Mmm, what comes to mind: the DRC-method uses (recorded) wave-samples to calculate the filter, right? So then it would be possible for me to (somehow) generate a wave as I would wish and use that for it (..thus would be able to use my own measurement methods and experimenting-possibilities)?
I'm sorry to not dig in yet and try the method et al, to get the answers on my own; but seen (some of) you have experience already, that could save time for me (in case it wouldn't give me what I need).
Something I'd also like to share:
My silent computer project:
http://www.prijsindex.net/cgi-bin/prijsindex.pl?lib=niels_projects_silent-computer
Im another thats been thinking about this for quite a while.
This is what im thinking of doing:
WinXP box (Envy42HT) -> Optical spdif -> Linux box (Envy24HT) -> BruteFIR (xovers & room EQ) -> Envy24HT -> I2S transmitter (TI MuxIT parts) -> Multiple chan DAC (with ASRCs) -> Gainclone amps
The Envy24HT in the Linux box will be heavily modifed for Optical input and the TI-MuxIT outputs.
Im not quite sure on the details of the DAC ill build yet.
Ill probably have fixed gain (no volume control) between the DAC and the Amps and use the Linux box (BruteFIR) to do volume control in 24bit.
Sound like a good plan?
The only thing im unsure about is if the copper connection between Linux box and DAC will cause noise issues... it would be nice to isolate the two somehow.
This is what im thinking of doing:
WinXP box (Envy42HT) -> Optical spdif -> Linux box (Envy24HT) -> BruteFIR (xovers & room EQ) -> Envy24HT -> I2S transmitter (TI MuxIT parts) -> Multiple chan DAC (with ASRCs) -> Gainclone amps
The Envy24HT in the Linux box will be heavily modifed for Optical input and the TI-MuxIT outputs.
Im not quite sure on the details of the DAC ill build yet.
Ill probably have fixed gain (no volume control) between the DAC and the Amps and use the Linux box (BruteFIR) to do volume control in 24bit.
Sound like a good plan?
The only thing im unsure about is if the copper connection between Linux box and DAC will cause noise issues... it would be nice to isolate the two somehow.
Nielsio,
You have to be joking about Microsoft not being reliable enough to use WMA. Microsoft is the largest software company in the world and have poured many millions into the development of WMA and it is the centrepiece of their media software strategy. Microsoft's codecs are recognised as some of the best around, and my reason for going with WMA is that the format has significant resources behind it, and you will start to see much wider support for it in hardware devices in the future (already you can get DVD players that will play WMA).
I just dont see the same sort of support behind Ogg (or FLAC, or APE etc) that there is with WMA. There are no real differences between the lossless codecs, and it is also possible to transcode between different codecs if needed without quality loss.
I agree with you about WM player, it is not designed for audiophiles, use Foobar or Winamp instead.
MWP,
What about a transformer or opticoupler between the Linux box and the DAC to isolate any PC interference coming in across the link. The muxit chips themselves must offer a degree of isolation especially if you get the grounding right.
Let me know how you go with your project. I want to try the RME soundcard approach first to sync the DAC clocks with a multichannel gainclone, then try a Panasonic XR-45 digital amp with the Muxit chips (as per Brian Brown's setup).
Regards,
Dean
You have to be joking about Microsoft not being reliable enough to use WMA. Microsoft is the largest software company in the world and have poured many millions into the development of WMA and it is the centrepiece of their media software strategy. Microsoft's codecs are recognised as some of the best around, and my reason for going with WMA is that the format has significant resources behind it, and you will start to see much wider support for it in hardware devices in the future (already you can get DVD players that will play WMA).
I just dont see the same sort of support behind Ogg (or FLAC, or APE etc) that there is with WMA. There are no real differences between the lossless codecs, and it is also possible to transcode between different codecs if needed without quality loss.
I agree with you about WM player, it is not designed for audiophiles, use Foobar or Winamp instead.
MWP,
What about a transformer or opticoupler between the Linux box and the DAC to isolate any PC interference coming in across the link. The muxit chips themselves must offer a degree of isolation especially if you get the grounding right.
Let me know how you go with your project. I want to try the RME soundcard approach first to sync the DAC clocks with a multichannel gainclone, then try a Panasonic XR-45 digital amp with the Muxit chips (as per Brian Brown's setup).
Regards,
Dean
deandob said:
Eeek... where do i begin?
Just because MS have money, doesnt mean they produce good products, just look at WinME!
IMHO WMA is a bad choice.
The format is copyrighted to hell (as everything MS is), and just really isnt as good as other formats that are avaliable.
Ogg Vorbis is 100% opensource, as are a lot of other formats which means they will have support and a good following for many years to come.
I wouldnt be supprised if in say 6 months, MS create another codec that better than WMA, and then say 1yr after that, all official support for WMA is dropped.
That just wont happen with a format such as Ogg Vorbis... there will always be people around in the opensource style community who will maintin it.
As for hardware support, Ogg is taking a while to become supported, but there are portable players now being sold with support.
With lossless codecs, you really cant go wrong.
If the CODEC becomes superseeded by another loosess format, just convert your music over... no quality loss... easy.
And with hard drive space as cheap as it is, going lossless isnt too expesnive anymore.
Hello !
Complicated !
Do you know you can transmit sound over Ethernet ? This way you don't need to add another soundcard ! I don't remember the name of the software which do this, but they exist. Look on the Net.
You'll still need an input in your Linux box to process the sound from your DVD/Home theater though.
ASRC is a useless, expensive and kludgey sonic compromise when it is so easy to synchronize the clocks.
You don't need to bother with that. SPDIF will go through insulation transformers. You can wind them yourself on a ferrite core or buy them. Copper SPDIF is OK with me as long as there are transformers. That's what I use.
Transformers as I said. Absolutely no problem, and saves you those expensive optical transceivers.
What is that ?
If you do that, don't use BruteFIR, simply use the volume control in your WinAMP or whatever...
Also a digital volume control is a highly compromized solution which will probably sound worse than a good chip digital attenuator.
Apart from PCM1704 I can see no DAC which can pretend to do 24 bits resolution (all the other are sigma-delta) and thus I would be very cautious of doing anything that may prevent them from using their full precision...
Have fun
Complicated !
Do you know you can transmit sound over Ethernet ? This way you don't need to add another soundcard ! I don't remember the name of the software which do this, but they exist. Look on the Net.
You'll still need an input in your Linux box to process the sound from your DVD/Home theater though.
ASRC is a useless, expensive and kludgey sonic compromise when it is so easy to synchronize the clocks.
You don't need to bother with that. SPDIF will go through insulation transformers. You can wind them yourself on a ferrite core or buy them. Copper SPDIF is OK with me as long as there are transformers. That's what I use.
Transformers as I said. Absolutely no problem, and saves you those expensive optical transceivers.
What is that ?
If you do that, don't use BruteFIR, simply use the volume control in your WinAMP or whatever...
Also a digital volume control is a highly compromized solution which will probably sound worse than a good chip digital attenuator.
Apart from PCM1704 I can see no DAC which can pretend to do 24 bits resolution (all the other are sigma-delta) and thus I would be very cautious of doing anything that may prevent them from using their full precision...
Have fun
From Windows to Windows and fro Linux to Linux, sure, its quite easy... but not from Windows to Linux.
The sound source will either be the WinXP box, or the Linux box itself (it stores music).
Hmmm, i think a lot of poeple would disagree with that.
But i want to use MuxIT so i can transmit all 6 channels using I2S over CAT5 without the I2S->SPDIF->I2S conversions.
I guess i can still use transformers, but they will need to handle ~500Mhz.
MuxIt is a system of data transmission ICs that Texas Instruments make.
So a search for MuxIT in these forums for details of a project already using the system.
Errr... i think ill use BruteFIRs...
I wont be using Winamp, and other audio sources may not have a volume control thats easy to use.
Maybe... 24bits should give plenty of room for volume.
And i dont want to build up a uC to control digital pots.
I know. Point is, if you can do without ASRC, it's better. ASRC si only a way to try to correct the flaws of SPDIF...
OK, then, buy a RME Digi96 soundcard with its 8 channel 24-96 ADAT optical output, and put an ADAT receiver in your DAC... Alesis makes those chips, they're cheaper than a SPDIF receiver; and you get 8 channels ! That's what I'd do.
Hi. Not sure if this is off the subject because you guy slost me a bit but I have a question.
I want to build a HTPC system to muse as a music jukebox (I'm going to just buy a few big hard drives and rip all my CDs to them uncompressed -- then put them in storage since I don't have the space).
Basically, I have a high-end external DAC and am looking for a great PCI card with digital output. I stumbled across the RME 96/8 and noticed the same stuff some of you did about the external clock option. I have a 5-pin i2s input (like the Audio Alchemy stuff used to have) on my DAC as well as coax SPDIF and AES/EBU.
Can I use the i2s input with the RME card? If so, how? will I need custom cabling? If so, any suggestions on getting the impendance right? Thanks.
Also, anyother suggestions for sound cards? Must work with Windows XP since that's what will probably run on my HTPC.
-Tom
I want to build a HTPC system to muse as a music jukebox (I'm going to just buy a few big hard drives and rip all my CDs to them uncompressed -- then put them in storage since I don't have the space).
Basically, I have a high-end external DAC and am looking for a great PCI card with digital output. I stumbled across the RME 96/8 and noticed the same stuff some of you did about the external clock option. I have a 5-pin i2s input (like the Audio Alchemy stuff used to have) on my DAC as well as coax SPDIF and AES/EBU.
Can I use the i2s input with the RME card? If so, how? will I need custom cabling? If so, any suggestions on getting the impendance right? Thanks.
Also, anyother suggestions for sound cards? Must work with Windows XP since that's what will probably run on my HTPC.
-Tom
Wait a minute - so how does the word clock thing work? I just snagged a cheap RME HDSP-9652 card and it has the word clock module built in. Rather than going through the SPDIF encode process, can I get synchronisation through a wordclock connection? Also, would I need additional components, or can I just use any DAC?
external high quality PC DAC
I think the way to do this is via USB. The TI chips are VERY cool. I ordered a sample of the PCM2707 today. It will regenerate (reclocked too! ) either SP/DIF OR I2S digital from USB audio. Use either to drive your DAC. I plan on feeding a TDA1541 or TDA 1543 directly running passive I/V and non-OS. Should be interesting....
cheers
Adam912
I think the way to do this is via USB. The TI chips are VERY cool. I ordered a sample of the PCM2707 today. It will regenerate (reclocked too! ) either SP/DIF OR I2S digital from USB audio. Use either to drive your DAC. I plan on feeding a TDA1541 or TDA 1543 directly running passive I/V and non-OS. Should be interesting....
cheers
Adam912
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