francolargo:
I am not sure we understand each other.
The compressed format you have was originally 44.1 or 48kHz?
If you uncompress, i.e. convert back to wav (b.wav), it will certainly sound different than the original a.wav. Both a.wav and b.wav will have the same bitrate of 2 x 16 x 44100 = 1.4Mbps. b.wav will have a limited frequency range since the MPEG-4 coder cuts high frequencies.
Now what I do not understand is why b.wav upsampled to 176.4 should sound worse than b.wav, while upsampling a.wav produces an improvement.
I am not sure we understand each other.
The compressed format you have was originally 44.1 or 48kHz?
If you uncompress, i.e. convert back to wav (b.wav), it will certainly sound different than the original a.wav. Both a.wav and b.wav will have the same bitrate of 2 x 16 x 44100 = 1.4Mbps. b.wav will have a limited frequency range since the MPEG-4 coder cuts high frequencies.
Now what I do not understand is why b.wav upsampled to 176.4 should sound worse than b.wav, while upsampling a.wav produces an improvement.
Hi Phofman,
What I know for sure is that the original compressed files are ripped at 256kb/s and play at 44.1kHz. I assumed that meant that during decoding, fewer samples were repeated in a serial fashion to fill up the bitrate to 1.4mb/s. That's the signal I don't want to upsample to 176.4 kHz.
Frank
What I know for sure is that the original compressed files are ripped at 256kb/s and play at 44.1kHz. I assumed that meant that during decoding, fewer samples were repeated in a serial fashion to fill up the bitrate to 1.4mb/s. That's the signal I don't want to upsample to 176.4 kHz.
Frank
Mid-thread conclusion please: Are we better off using something like an Empirical Audio USB to Spdif converter, or something like a squeezebox duet (assuming both are run into a decent quality DAC)?
http://www.empiricalaudio.com/frComputer_Audio.html
http://www.slimdevices.com/pi_duet.html
http://www.empiricalaudio.com/frComputer_Audio.html
http://www.slimdevices.com/pi_duet.html
francolargo said:
The MPEG compression works rather differently. Basically, it performs frequency analysis of the incoming wave, breaks the whole band into many sub-bands and records at regular intervals the sub-bands signal levels. Lower bitrate means information only on more significant frequency sub-bands is included in the resultant stream.
The decoder kind of "plays" the frequencies, creating a new waveform (wav) somewhat similar to the original one. The new waveform does not repeat samples in a serial fashion, it just has a different shape from the original wav. Since it is a regular wav (PCM), its upsampling should produce the same improvement (if it is the case) as upsampling the original wav.
Interesting!
Perhaps with the mpegs I'm noticing an effect similar to one noted over on the ESS Sabre reference thread.
http://www.diyaudio.com/forums/showthread.php?postid=1735012#post1735012
There it was observed that with a better DAC the poorer quality recordings actually sound worse. We ARE talking about subjective SQ, and in a similar way, perhaps the degradation of mpeg encode/decode is resolution dependent. I was not planning to re-rip most of my 'average'-quality recordings to a higher resolution but now am re-evaluating... It may be necessary to maintain two source resolutions, a lossless one for the living room and a compact one for the iPod...
over-and-out,
Frank in Mpls.
Perhaps with the mpegs I'm noticing an effect similar to one noted over on the ESS Sabre reference thread.
http://www.diyaudio.com/forums/showthread.php?postid=1735012#post1735012
There it was observed that with a better DAC the poorer quality recordings actually sound worse. We ARE talking about subjective SQ, and in a similar way, perhaps the degradation of mpeg encode/decode is resolution dependent. I was not planning to re-rip most of my 'average'-quality recordings to a higher resolution but now am re-evaluating... It may be necessary to maintain two source resolutions, a lossless one for the living room and a compact one for the iPod...
over-and-out,
Frank in Mpls.
phofman said:Oops I forgot this was not the linux threadPerhaps someone has produced a similar functionality for windows.
You mean like the basic transcoding feature with most popular media players like Windows Media Player, iTunes, J.R.Media Center, Foobar2000, Media Monkey, etc., etc., etc.
All of these apps will automatically transcode your audio on-the-fly when you transfer tracks to a portable player like the iPod.
greggp said:You mean like the basic transcoding feature with most popular media players like Windows Media Player, iTunes, J.R.Media Center, Foobar2000, Media Monkey, etc., etc., etc.
All of these apps will automatically transcode your audio on-the-fly when you transfer tracks to a portable player like the iPod.
Thanks for the insight. The virtual filesystem mp3fs does it slightly differently. It provides a read-only directory where you get a mirror of a master directory structure, and any flac file in the master is automatically converted to mp3 when the mp3 file gets read. It is a standard filesystem directory structure, you can access it with any program or tool.
But in the end, all the alternatives do the job
Re: Re: i2s
Hey Frank,
Would this (the pci card from the maxio 032) work in the same way? I mean i2s out and wordclock in using the serial breakout connector?
The vista drivers are not available yet but they say they will be soon (only difference vs the maxio 032e is the interface)
http://namm.harmony-central.com/WNAMM05/Content/ESI/PR/MaXiO-032-PCI-lg.jpg
http://www.esi-audio.com/products/maxio032/
francolargo said:
Hey Frank,
Would this (the pci card from the maxio 032) work in the same way? I mean i2s out and wordclock in using the serial breakout connector?
The vista drivers are not available yet but they say they will be soon (only difference vs the maxio 032e is the interface)
http://namm.harmony-central.com/WNAMM05/Content/ESI/PR/MaXiO-032-PCI-lg.jpg
http://www.esi-audio.com/products/maxio032/
Re: Re: Re: i2s
Hi Telstar,
I have no direct experience with that equipment but did look at the manual and photos online. It appears that this card is significantly different from the one operating the 1010e. The main interface to the breakout box is a firewire-type connector, each of the 4 connectors manages 8 channels (supporting up to 4 boxes). The manual says that the digital signal protocol is 'exclusive' to ESI and they have named it "E.D.I". The other connector on the card is for MIDI and s/pdif I/O, and the spec mentioned that the maximum s/pdif frequency on this connector is 96kHz. I imagine that the firewire cables transmit I2S and that it wouldn't be too difficult to find the I2S signal on the board of the breakout box. But if you pulled the I2S from the break-out board it also looks as if you would either a) have a major effort optimizing and configuring a new DAC - there is probably next to zero free space, or b) have to abandon most of the other capabilities of the breakout box. [Disclaimer: I'm certainly not an engineer!] We can't come close to guessing what happens to the signal before the firewire output because that area is sandwiched and we can't ID any chips from the photo. The main signal processor of the PCI version of the board appears different from the Envy 24HT chip on the 1010e, though it is blanked out on the photo. However, the photo of the PCIe version of the board shows a Xilinx spartan chip and the pins of that chip look similar to the one on the older PCI board. So, I guess the question is: What is the signal protocol that ESI has named E.D.I.?
Frank in Mpls.
Telstar said:
Hi Telstar,
I have no direct experience with that equipment but did look at the manual and photos online. It appears that this card is significantly different from the one operating the 1010e. The main interface to the breakout box is a firewire-type connector, each of the 4 connectors manages 8 channels (supporting up to 4 boxes). The manual says that the digital signal protocol is 'exclusive' to ESI and they have named it "E.D.I". The other connector on the card is for MIDI and s/pdif I/O, and the spec mentioned that the maximum s/pdif frequency on this connector is 96kHz. I imagine that the firewire cables transmit I2S and that it wouldn't be too difficult to find the I2S signal on the board of the breakout box. But if you pulled the I2S from the break-out board it also looks as if you would either a) have a major effort optimizing and configuring a new DAC - there is probably next to zero free space, or b) have to abandon most of the other capabilities of the breakout box. [Disclaimer: I'm certainly not an engineer!] We can't come close to guessing what happens to the signal before the firewire output because that area is sandwiched and we can't ID any chips from the photo. The main signal processor of the PCI version of the board appears different from the Envy 24HT chip on the 1010e, though it is blanked out on the photo. However, the photo of the PCIe version of the board shows a Xilinx spartan chip and the pins of that chip look similar to the one on the older PCI board. So, I guess the question is: What is the signal protocol that ESI has named E.D.I.?
Frank in Mpls.
Audio Alchemy powered digital cable...
... was the EDI (electronic digital interface). I believe it took an spdif signal broke down into i2s and trasmitted to a receiver which converted back to spdif (via bnc) to plug into a dac. Question: is my recollection correct on edi--> spdif--> i2s-->spdif? Also does anybody have a schematic of the transmitter/receiver? Reason is I have an EDI and would like to hack i2s out of my juli@ card into the transmitter if possible. Appreciate any input on this.
... was the EDI (electronic digital interface). I believe it took an spdif signal broke down into i2s and trasmitted to a receiver which converted back to spdif (via bnc) to plug into a dac. Question: is my recollection correct on edi--> spdif--> i2s-->spdif? Also does anybody have a schematic of the transmitter/receiver? Reason is I have an EDI and would like to hack i2s out of my juli@ card into the transmitter if possible. Appreciate any input on this.
Re: Audio Alchemy powered digital cable...
You're in luck:
http://www.diyaudio.com/forums/showthread.php?s=&threadid=102851&highlight=
Frank in Mpls.
theob said:
You're in luck:
http://www.diyaudio.com/forums/showthread.php?s=&threadid=102851&highlight=
Frank in Mpls.
I have bought an E-Mu 0202 USB audio interface and measured the M0, M1, M2, M3 pin of the onboard CS4392 DAC. M1/M0 measure "01" (24 bit I2S according to datasheet) when foobar is playing 16/44.1, 24/88.2, 24/96, 24/192 flac files. I think the chain is like this:
foobar -> E-Mu ASIO driver -> Philips USB interface ic (bulk mode) -> E-Mu's FPGA -> DAC
I think the FPGA is the master clock as there are two smd crystals. One 22.579M and the second one is 24.576M. It also converts everything to 24 bit I2S.
Anyone has tried tapping this I2S to external dac?
foobar -> E-Mu ASIO driver -> Philips USB interface ic (bulk mode) -> E-Mu's FPGA -> DAC
I think the FPGA is the master clock as there are two smd crystals. One 22.579M and the second one is 24.576M. It also converts everything to 24 bit I2S.
Anyone has tried tapping this I2S to external dac?
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