anoutsos

My latest attempt was a few months ago. I extracted the SACD as a single DFF (Master), together with the CUE file, using sacd_extract. Then, I converted the DFF file to AIFF 88.2/24 using Saracon, and finally I used the CUE file to split the AIFF file into tracks, using XLD. This process creates only two clicks, one at the beginning of the first track and one at the end of the last. It is very easy to remove these clicks manually, using software like Sound Studio, etc, but I did not bother since those clicks are only particularly annoying when they occur between tracks. Cheers, Aris.

I just want to clarify that I am not interested in the DSD streams at all. I just want to have the album tracks of the SACD as stereo PCM 88.2kHz/24bit files. The metadata that so many people are worried about can be inserted at a later stage; it's not an issue. I would also like to convert the DSD to PCM using Saracon, because it removes all the useless ultrasonic information above 88KHz. So, I really like mansr's idea of extracting single DFFs from the ISOs, converting those to PCM and then splitting them with a cue file. I have not tried that yet, but it sounds very promising.

I am planning to use sox to trim 0.0006 s from the beginning and end of the audio files (where the chirps are always found). My command is sox input output trim 0.0006 =(length-0.0006) Can anyone see any problems with this approach?

Thank you for your reply. Does the latest ISO2DSD for MacOS X use sacd_extract v0.3.8 or v0.3.6? As for the format, I have been extracting DFFs (converted to DSD), because Saracon does not recognise DSFs. I remember that at some point it was being suggested that the solution to avoiding the clicks was to first extract as DFF and then convert to DSF. I don't think that this is valid, though, since I have been extracting directly to DFF and I still get the clicks. By the way, if I am only interested in DFFs, would I not get the same output by using the "-p -c" flags in sacd_extract? As far as ID3 tags are concerned, I typically add them later anyway. sacd_extract parses the names of the tracks and that's enough for me. Actually, I've just compared v0.3.6 to the one supplied with ISO2DSD. The former works twice as fast! So, there must be quite some difference.

So, the spike is a feature of the conversion from DSD to PCM, but there are players, like A+, that suppress it? Does the ISO to DSD extraction introduce this spike, too? Also, if part of the data flow is suppressed by A+, how does A+ reproduce gapless recordings? Does this introduce gaps during playback? I also wanted to ask about the difference between Bogi's ISO2DSD and using sacd_extract from the command line. Do they lead to exactly the same output? I am asking because I would rather script sacd_extract for batch jobs than load them into a clunky graphical JAVA interface. Thanks.

Oh, I see. I also used ISO2DSD, but I thought that the spike is introduced at the next step, when one converts DSD to PCM. I have read somewhere (“Pop” goes DSD? Why does this happen?) that the pop happens because of some sort of levels normalisation. However, the point remains that there is a spike in the data stream and it would be good to get rid of it at the extraction or conversion stage. Also, if A+ and other players suppress those spikes, does it mean that they are not bit-perfect or are they selectively alter only that particular feature of the data stream?

I am using Amarra (v3.0.3), in playlist mode. To iron out the spike at the beginning of the tracks I use Sound Studio. I have noticed that after eliminating the spike and saving, the AIFF file is different by a few kb. Could I be screwing up the bit-stream by messing with it like that? The interpolation is applied very locally, over a few tens of microseconds around the spike.

Hi all, I am new here and I have read most of this thread. I have used Weiss Saracon to convert DSD to AIFF (PCM) and I get the infamous pops at the beginning of tracks (see image). I have tried the simple method of selecting the offending spike and applying interpolation, which removes it. I am very close to eliminating this spike from all my DSD converted songs, but it may take the rest of the year to do it. Nevertheless, is there an automated/scripted way of doing this? The current procedure is "Open AIFF", "Zoom into the beginning of the track", "Selecting the spike", "Applying interpolation" and "Saving". I have 1,613 songs, so it would take a hell of a long time, but I am almost crazy enough to do it (then back up the songs thrice!). I have a Mac, by the way. Apart from the expensive software that people have suggested, is there another way of eliminating the spike? Thank you all in advance. Aris.

Thanks for your reply, David. Perhaps I was not clear, though. I have already installed the HDD as a second drive _inside_ the Mac Mini, using the dual drive kit, so there is no space for additional components. The masking tape sounds interesting but I am afraid to wrap the drive with it. Also, I have discovered that "pmset" can control the duration of inactivity before the drive spins down. Finally, I am not convinced that the noise is due to vibration, but I will open the MM again and check whether it's touching e.g. the WiFi grill sitting just on top. Thanks, Aris.

I would like some advice. I have set up a Mac Mini as my music server. I have installed an SSD for the OS and I have all my music on a second HDD. Yesterday, I upgraded the HDD to a Samsung 2TB disk (the original drive was a 500GB Hitachi). Unfortunately, I can now hear the Samsung drive spinning. It is supposed to be silent but it seems to be not as silent as the Hitachi. My question is: Can I set the HDD to spin down frequently, so that I can load the tracks on to the memory and listen to my music without the background noise of the HDD? Thanks, Aris.

I would like to set the default audio device for Amarra (optical) to be different from the system output (HDMI), so that I can use the HDMI for video and audio on a different input on my AV processor. This way I can keep one input on my processor purely for music and use the other input for movies, etc. Is this a good idea and will Amarra stick to the optical device or will it detect that the HDMI connection is active and try to revert to that. I have "Follow midi settings" unchecked. I am worried that although I have saved the configuration in Amarra, it will revert if it sees the default system device. Thanks, Aris.

I have been experimenting with playing back my music via the built-in USB port on my Marantz AV8801 and comparing that to playback with Amarra 3.0.3 connected to the AV8801 via TOSLINK (optical). Can anyone tell me if Amarra's main responsibility is to send bits to the AV8801 accurately and with the correct timing? Does it also perform any sort of modification to the bit stream toward improving the sound (I have EQ turned off)? I have not been able to tell the difference between Amarra and the USB playback, so I was wondering what the advantages are of using a computer and Amarra instead of a USB drive or a NAS connected to a router, connected to the AV8801 via RJ-45. Thanks, Aris.

Hi, I am new to A+ (v2.2) and I am trying to decide between multichannel DSF and DFF (DSDIFF) for my hi-res A+ library. I have heard that DFF does not support ID3 tagging. However, I have been able to make changes to the information of DFF file in both A+ and Yate, and those changes stuck with the files when viewed by the alternative application. How is it then that DFF files do not support metadata? The reason I ask is because I would prefer to have the much smaller DFF files instead. The other option would be to convert the DSF files to flac (24/88.2), but that would require some decisions about the gain and filtering between DSD and PCM. Thanks for the help. Aris.

I was too quick to make my conclusion. I tried again DSD-->FLAC-->ALAC and this time it didn't work. So I guess Amarra has a problem with ALAC from XLD, no matter what method is used to get there. The iTunes method works, though. Cheers, Aris.

OK, I have now tried both methods. Ralph's suggestion works perfectly. The method via another format in XLD gives variable results: if AIFF or WAV is used between DSD and ALAC, then the final ALAC does not play, as in the case of direct conversion between DSD and ALAC in XLD. However, if FLAC is used (with the default XLD settings) between DSD and ALAC, then the final ALAC plays perfectly. So, it seems that I may need to start converting my files into FLAC first and then into ALAC. It has to be said, though, that 99% of my conversions directly to ALAC in XLD (including several DSD albums) work fine. (Come to think of it, many have been from FLAC! But not all.) Hence, I am a bit hesitant to adopt the two-step solution on a regular basis. Perhaps it is better to stick with the direct conversion to ALAC and whenever I see a problem switch to the other method. Anyway, thank you all for your suggestions. Of course, if someone can point me towards another piece of software that would convert directly to Amarra-compatible ALAC from any format, I would be very interested. Cheers, Aris.