dex Otaku

http://forums.minidisc.org/index.php?showtopic=14173 Note that I have been listening to the encoded content from SS 3.4 and there are no balance issues with it. You appear to be bent on making out SS 3.4 as something totally broken, when in fact many of us have no problems with it whatsoever. You also appear to be the only person experiencing the balance issue you've mentioned, however, you haven't updated your own thread with more info as requested [and required for others to help in any way], and you don't seem to be interested in following the suggestions we've made for diagnosing the exact source of the problem. Further, you've been blaming the issue on regionalisation, which is completely specious. My experience on this board since mid-2004 has suggested that about 90% of SS problems are actually caused by users who don't maintain their OS and other software well. As I said already, there are many of us out here who have none of these issues - but then, most of us who profess this openly maintain our OS and other software fairly rigourously, if not in fact rigidly. Please, go back to the thread you started, and if you actually want more results, follow the suggestions we've made. Otherwise, please stop adding any further arguments to the mix, and don't give people advice based on something that you evidently won't explore any further yourself.

1st-generation HiMD players do not natively support MP3 playback, so no, you can't download MP3s to your NH1 without transcoding to atrac3plus. This is a limitation of the hardware, not SS.

You can use SS with multiple HiMD units without difficulty. I do. No. I'd stil advise, as I normally do, backing up your library before upgrading just in case you run into any problems. No. I've used Asia-Pacific, Canadian, European, and US versions of SS to upgrade previous versions without any problems in the past. My current version is SS 3.4 European, upgraded from SS 3.3 US. The only difference I've ever noted between the regional versions is where the Connect store points to [or in the case of the Asia-Pacific version, the fact that it isn't present]. If you use the Connect store, wait for the version specific to your region. If not, don't worry about it.

Not to repeat myself, but since it appears you didn't read it:

To the best of my knowledge, there's no way of recovering the lost information on your discs. You might try using some commercial data-recovery programs on the discs themselves .. but in all honesty, your data is probably just lost. You could also try contacting Sony to see what potential solutions they can suggest. Sorry for your loss, in any case. Perhaps the lesson here is that backups are a good thing.

Quick answers: * if you have a european/british unit, you might be able to uncap the volume restriction, though this involves a certain amount of risk to the unit [going into test mode and changing firmware settings, if the unit supports it] * get more sensitive 'phones as greenmachine suggested * buy or build an external headphone amp to provide more power

Hint: read the thread you just posted in. What it boils down to is that when recording using line-in there is no way to turn auto t.mark off.

I'm really beginning to wonder here .. there's the HiMD uploading FAQ, and there are threads stickied in the relevant fora about how to "upload" MD recordings .. and no one seems to read them. In the past week alone I've seen several users ask this same question. What's the problem, here?

A further note to add - 1GB discs are faster both in read and write speeds than legacy MDs. Take a look at the HiMD FAQ at www.minidisc.org for specifics.

Sure there are, but those geeks would have to crack and modify the firmware in all the recorders, not just the modules used by SS.

Why don't we go back to the beginning, shall we? Post your system specs according to the forum rules, and most specifically, the module versions being used by SS, as listed in help->about. To my knowledge the only difference in the release versions of SS from different regions is where the Connect store points to [which can be changed by editing the URL in SS's registry settings]. The module versions should be identical in each region. Also: * what is the source of the tracks you're encoding? CD? MP3s? You haven't specified. * does the balance problem occur when playing in SS as well as on your portable? You haven't specified. * if yes to the above, have you tried downloading HiMDRenderer and converting one of those mono files back to WAV, and comparing channel levels? Or for that matter, uploadinga short chunk of the resulting WAV [or better, FLAC] file so that we can check it out? * how about sending one of us one of your mono SS-encoded files so we can compare them ourselves [being sure to encode without copy protection]?

Thanks raintheory.. one less thing to try out myself.

Related news: SS 3.4 u/l and d/l testing, with special notes on MP3 conversion.

And for further "control" .. use HiMDRenderer to convert the tracks back to WAV afterwards and see what they look like. Mono input should give output with two perfectly-matched channels, level-wise.

See here: SS 3.4 u/l and d/l testing, with special notes on MP3 conversion.

In the thread "SS 3.4 upload woes?" by kurisu, I mentioned testing uploads with SS 3.4. Given the oddities some users have been experiencing with downloads as well, I've also decided to see if I can reproduce their errors. Please note that all testing I'm doing is with HiMD mode discs only. netMD mode uses a separate module and is bound to have issues that are completely separate as a result. Uploading tests * uploading of unmodified tracks in HiLP, HiSP, and PCM modes <blockquote>HiLP, HiSP, and PCM uploads from both my RH10 and NH700 work as expected. In particular, as some users have noted having problems with PCM uploads, I made multiple test recordings with both units using both line and mic in, on both a HiMD-formatted MD80 and a 1GB disc. No problems experienced on any count.</blockquote> * handling of tracks with known write errors [known problem] <blockquote>Unexpected results here - uploading went straight through write errors with no problems at all. The resulting tracks have clicks where the errors were. Tracks are playable on the unit both beforehand and after. Assuming it always acts this way [which implies that error-recovery has been vastly improved], this is a major change since previous versions with their seemingly random behaviour [including erasing the tracks, erasing the tracks and all subsequent ones on the disc, skipping the track and rendering it unplayable, and other nasties].</blockquote> * handling of very short track lengths [known problem] <blockquote>Tested with HiLP,HiSP, and PCM modes. Track lengths of 1 second [i.e. bouncing my finger on the T.MARK key repeatedly] upload and play properly. I haven't tried combining them yet, but since they do play, I'm assuming for now that they likely will do so properly. I could be wrong, though. If it's important enough to anyone else, please try for yourself to see what results you get. Previous versions would create blank tracks in the library after upload if they were this short on the disc, attempts to convert to WAV would leave 0-length WAV files, and attempts to combine [at least with 3.1] would sometimes result in an error/abort. Another improvement, though obviously not as critical as the write error issue.</blockquote> * handling of contiguous tracks recorded via line-in and automatically trackmarked by the unit [specifically, whether combine still has the repeated section issue, which is admittedly likely a hardware problem] <blockquote>Same results as before, as expected. This is a hardware issue and would require seeking through track ends/beginnings [like HiMDRenderer does] in order to be fixed. The easiest solution is still not to combine recordings made via line-in with SS, rather to use other editing software afterwards.</blockquote> Downloading/encoding tests A note before everything else: downloading compressed tracks with SS 3.4 to either my NH700 or RH10 appears to be significantly slower than it was with SS 3.3. I'd estimate that it takes at least 50% longer for a given track than it used to, and there's no difference between downloads of MP3 or a3/a3+ tracks of the same bitrate. Needless to say, this seems odd to me. PCM downloads and uploads of any type do not appear to be any slower than before. Also, please note that MP3s of any sampling rate other than 44.1kHz have to be resampled and transcoded [one way or another] before they can be used with HiMD. While yes, this information is in the manual for MP3-capable HiMD units, the most prevalent reference is on the technical specs page and not anywhere that the average user would look. Hence my mentioning it here, since audiobook users tend to gripe about SS's poor handling of their [already specified-as] non-compliant files. Attempting to download a 48kHz MP3 directly results in the following message from SS: MP3s of any sampling rate other than 44.1kHz transferred directly to any MP3-capable HiMD will only result in the message "CANNOT PLAY" on the unit when playback is attempted. There's [a lot] more relating to this at the bottom under "MP3 transcoding." * downloading single tracks <blockquote>Tested with: MP3 @ 192 and 320kbps, as well as VBR files; a3+ @ 192, 256, and 352kbps, also PCM. No problems experienced regardless of file format [assuming they're compliant].</blockquote> * downloading multiple groups <blockquote>Several users have expressed having difficulty doing this with both SS 3.3 and 3.4 already. Tested with: MP3 @ 192 and 320kbps, as well as VBR files; a3+ @ 192, 256, and 352kbps, also PCM. No problems experienced regardless of file format [assuming they're compliant], with as many as 10 groups/albums at a time.</blockquote> * encoding from CD or WAV files <blockquote>I tried this with 192 and 352kbps a3+ modes, with "high quality" option enabled. No problems whatsoever experienced. In particular, no misplaced/moved trackmarks [tested with self-made mix CDs with segueing tracks and Pink Floyd's Dark Side of the Moon] and no level differences between channels. I'll note that encoding from both 16- and 24-bit WAV files @ 44.1kHz and 48kHz sampling rates works fine. See below for the reasons why I verified this.</blockquote> * MP3 transcoding <blockquote>Lots of issues here. This is probably the single most important downloading-related issue for netMD and HiMD users, so please read the whole thing. I digress! Sony have made me a liar by downgrading SS! Using FFDshow with 24-bit output, SS 3.4 ends up with 0-length OMA tracks. This is very odd, because it used to work just fine this way, though admittedly I haven't transcoded any MP3s with SS since v3.1. Using FFDshow with 16-bit output, everything works. Almost. From 44.1kHz sampling-rate files, SS transcodes just fine. From 48kHz files [which most will never encounter] the transcoded output is wracked with severe aliasing distortion. Changing the active codec on my system to FhG's [as well as switching between libmad and mp3lib in FFDshow] does not affect this; the aliasing distortion is still present, and HIGHLY audible in either case. To those who don't know what aliasing distortion is, it sounds like high-pitched [harmonic] buzzing. This also seems odd to me, because I had transcoded 48kHz MP3s with SS in the past [with FFDshow in 24-bit mode, no less] with no ill effect. It's even more odd to me, since encoding 48kHz WAV files, which also get resampled, works perfectly fine. For all intents and purposes, it's safe to say that at this stage SS can ONLY accept 16-bit, 44.1kHz input from any MP3 codec. Whatever resampling method SS is using internally is obviously NOT working at all when the data is coming from a directshow MP3 codec. As SS has already been known to have problems - regardless of what MP3 codec you're using - with sampling rates other than 44.1kHz, I can only recommend to anyone wanting to transcode MP3s in any rate other than 44.1kHz to resample the tracks in question before attempting to do anything with them in SS. I will suggest two ways to do this: [though these are by no means the only ways possible] <blockquote>* The first is to decode the non-MD/HiMD-compliant files to WAV and upsample them to 44.1kHz with an editor such as Audacity, or the built-in functions of conversion software such as dBPowerAmp BEFORE doing anything with them in SS. If using editing software, this can be done with mono WAV files, making them 1/2 the size, btw. * The second is to use FFDshow as your MP3 codec. Note that I do not really advise this for non-technical users as the myriad options it presents may end up being little more than confusing to them. FFDshow has an internal resampling filter, meaning you can set FFDshow to resample to 44.1kHz and use 16-bit output only, then transcode MP3s of any sampling rate in a single step without having to resort to WAV files, external editors, or other conversion software at all. For users with collections of audiobooks, this may be the most attractive method to convert your low-bitrate, low-sampling rate [i.e. non-MD/HiMD-compliant] MP3s in a single step without having to worry about disc space and/or extra processing time. Note that if you do use FFDshow as your system MP3 codec, and choose to do this for any reason, you should disable resampling when you're done. Normal listening should not involve resampling [particularly since AVI files with interleaved MP3 soundtracks @ 48kHz will experience a performance hit due to the resampler if it's on when it doesn't need to be.].</blockquote></blockquote> ------------------ I haven't gotten to these yet, but will post again in this thread when I do: * handling of tracks that have been edited on the unit [known problem] * exact behaviour of SS regarding attempts to upload non-DRM'd tracks from different sources * for sh*ts and giggles, seeing if split WAV files that comply to 75fps lengths remain gapless after encoding to a3+

As documented elsewhere on the forum countless times since mid-2004, yes, PCM transfers are not altogether speedy. The read speed of HiMD from a 1GB disc peaks at around 9Mbps, less than the top speed of USB 1.1. If 100% of that speed were used for PCM [1.44Mbps] uploading, you would get a speed of 6x realtime. Real-world usage, and this has been the same since SS 2.1, has PCM uploading at between 2-3x realtime, which would put your 10 minute upload for 30 minutes of recording at the high end of the normal range, speed-wise.

Actually, all atrac3plus modes use m/s joint-stereo encoding. The two channels it encodes are mid/side, the sum and difference of the left and right channels. That said, any differences in levels between normal stereo channels would have to be taking place either before the a3+ encoder [since differences caused by the encoder itself would manifest as either the middle or sides being louder than normal, not one side being louder than the other] or during decoding in the player. I am in the midst of doing some tests with SS 3.4 to see if I can reproduce any of the oddities people are bringing up, and so far, nothing has been reproducable. Everything is working normally. Test encodes done so far: a3+ 192, 256, and 352kbps, "high quality" option turned on, ripped directly from CD. Chris G: You should try encoding something that you know is dual-mono [exactly the same signal in both channels, CDs mastered this way are usually identified with a MONO label on the cover, examples include some early Beatles and Beach Boys recordings] to see if the balance problem happens with that material. If you lack any CDs like that, just make a mono recording and encode it with SS, or take something you're familiar with and use whatever editing software you prefer to convert it to mono, then encode with SS. Please let us know the results.

Doing upload tests with 3.4 - so far, no problems whatsoever, including with PCM mode, from either my RH10 or NH700. These problems must be something local to your installations. Not to say that it isn't weird as hell.

Every optical drive has a slightly different read and write offset for audio. Programs like Exact Audio Copy can correct this by checking a drive/firmware version database or by using reference discs. Most ripping software do not support this function, so whether your rips have accurate positioning is completely at the mercy of your drive/software combination. Read positioning can also change from rip to rip without correction, and can vary depending on whether you rip an entire disc vs. a single track. It might help to turn on the "error correction" mode that SS supports, which warns that it can slow ripping down. This probably uses more accurate positioning. In all honesty, I've turned that option on and left it on since first installing SS 2.1 in mid-2004; I've never run the program with it turned off, and I've never had a problem with a rip that wasn't caused by a faulty disc.

Edit: I forgot one of the most interesting things Fb2k can do, which I have occasionally used. * the ability to play back files of any supported format inside a .RAR or .ZIP archive just by dropping the archive on a playlist; no unpacking required.

Yes and no. Fb2k does decoding/playback at whatever bit depth you want, with decent dither, for one. This means if you use EQ or any DSP [which usually use >24bit resolution for processing and downconvert to the output bit-depth] the quality doesn't suffer significantly. I'll note that probably 90% of people won't notice the difference anyway, but on some material I do. Fb2k's interface can be customised to show what you want it to. Its library management tools lack the database abilities of iTunes, Winamp, or even SS, but I organise things in a way that the structure on my hdd itself is sensible and makes things easy to find to begin with, so that doesn't matter so much to me. The way I've customised the interface is basically to look something like the old Winamp 2.x [pre- nullsoft library manager] days, which even with a fairly large library I find easier to manage. Personally, I wish it had an iTunes'-like interface, which I find the most sensible combination of library database and playlists. My main reasons for using Fb2k: * support for all the formats I use, including FLAC, WavPack, OGG Vorbis, MP3, MPC, and others * support for surround formats including DTS and AC3 * ID3v2 and APE [which I don't use] tag support on basically any format except WAV [yes, I can tag DTS tracks] * freeDB support built-in: select a range of tracks [i.e. an album], look it up, tag the tracks right in the playlist * mass rename/move function that is programmable to suit how you like to organise things; I love this function in particular because I can rename/copy/move my entire library in a single step, conforming all tracks to the same structure, to match whatever I want - my own naming scheme, one like iTunes', or one with short names for use with home DVD players limited to 8.3 filenames, and it does so very quickly. * the ability to convert/transcode to most popular formats just by selecting tracks, right-clicking, hitting "convert" in the menu, and telling it what codec, along with the ability to tell it what naming scheme to use when doing so [i find this especially useful for APE downloads, because the APE format is so thoroughly unreliable - having no error-correction built in - that the first thing I ever do with them is transcode to WavPack] * almost everything about it is programmable by the user * small memory footprint, no required support of SQL or anything ridiculously cumbersome just for a music library * ability to [via DSP plugins] downmix surround recordings and transcode for playback on portables in a single step * the EQ it comes with is much better than either Winamp's or iTunes', giving slightly more flexibility for tuning room resonances [+14dB peak at 100Hz to cancel out, anyone?] .. though really, a decent Q-filter plugin would be nice * true gapless playback of any tracks in any format that comply with CD's 75fps; lossless formats, DTS, AC3, newer lame-encoded MP3s, &c. play with no gaps whatsoever, without using crossfading, edge-detection, or any form of DSP to do so * it's rock-stable [i've never had it crash. Ever.] * with some sound cards [my revo 7.1 not included] it supports unprocessed SP/DIF output including AC3, DTS, and MP2 directly as well as PCM All that said .. the new beta breaks all compatibility with current plugins from what I've read [on the official site], so hopefully everything that's in the one I'm currently using [the current stable build with extras] gets updated once it gets released as more than beta. I was a Winamp user since the pre-1.0 days. I tried out iTunes the week it came out for PC. I've also tried MusicMatch [a piece of unmitigated crap that I will never touch again], WiMP [most confusing interface of any player ever made, makes even SS look good in terms of usability], and of course SS [which I will never use as a player/library manager for anything other than uploading/downloading because it's such a behemoth]. Fb2k is definitely not for everyone, but if, like me, you deal with myriad different formats, prefer to keep things in their original format unless you must send them to a different medium, need to tag tracks in various formats, often need to copy large ranges of tracks to other naming conventions, &c. &c. ... then this is really the best tool I've found that has them all under one interface [albeit the default is ugly as hell]. What it lacks in terms of prettiness, it makes up for in truly useful functions that nothing else has. In an ideal world, a fantastic library manager would be married to all the other great stuff, and this would become the single most versatile audio software available.

No really. Don't play with me. Tom Ellard / Severed Heads [is/are] among the reasons why I do some of what I do today. I'm sitting here with the "BIGOT BOOKLET" giggling right now.

Hmm. Can't say that I was paying attention to this .. for two reasons - first, I don't use netMD mode on any discs, and second, LP4 sounds like poop. Now that you've mentioned it, though, yes - it's completely gone from 3.4. In fact, the distinction between atrac3 and atrac3plus is gone .. the list of possible encoding modes is either ATRAC or ATRAC advanced lossless now.. hmm. Theories: * Sony have determined by whatever means that LP4 is not often used and its inclusion is redundant * Sony have realised how bad the quality of LP4 is compared to virtually any recent codec at near the same bitrate * Sony have deemed the netMD market so tiny compared to the hdd, flash, and HiMD markets combined [all of whice support atrac3plus] that they have removed the codec because, well, it's redundant and has been superceded by a3+ bitrates * Sony have removed LP4 support to push netMD customers into upgrading to equipment that is a3+ compatible Personal opinion? Some of all of the above. This could also imply, since the vast majority of the netMD / MDLP market is in Japan, that they're trying to push the Japanese market into dropping their old equipment in favour of Sony's newer offerings.

Foobar2000.