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dex Otaku

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Everything posted by dex Otaku

  1. dex Otaku


    Going on hiatus again. If the board is lucky, I will never return.
  2. Sounds like you're encoding to HiSP [256kbps], which gives a usable 7:55 recording time. Using HiMD's own audio format [atrac3plus] you can also encode to: LPCM [uncompressed, 94 minutes] 352kbps [just under 6 hours] 192kbps [just under 11 hours] 64kbps [aka HiLP 64] [around 33 hours] 48kbps [aka HiLP 48] [around 42 hours] What you deem to be the minimum acceptable quality is up to your ears. Check SS's encoding options, try encoding to each bitrate, and see what you prefer. Choose your own compromise between time vs. quality.
  3. http://forums.minidisc.org/index.php?showtopic=14222
  4. Just out of curiosity, have you not switched the menu language to your own [or at least, one of those available that you understand better than Japanese]?
  5. Man I wish I could gets them that cheap here [where they're almost 4x that price].
  6. When you import your MP3s into SS's library, it reads the existing tags. If you keep well-tagged MP3s [unlike most people] then the results it imports, and subsequently tags the same tracks with when they're downloaded to a player, are quite readable. The difference is basically made by whether you keep your tags orderly or not. SS does have limitations relating to the length of individual tag fields [like truncating stupidly long song titles]. There as also a limit to how much text you can send to the player [rather, a specific disc] in tags as well, but I believe that for the 1GB discs t
  7. There is very limited support for on-unit playlisting [by using bookmarks]. On the other hand, you can divide the tracks you're downloading into groups [i.e. "for driving", "for chilling out" &c.] using SS. SS has playlist tools [you could make a playlist for driving and download it into a single group called "for driving"], and while they serve a basic purpose SS lacks a few features [like being able to randomise the playlist before downloading] of other library software. SS also lets you import M3U playlists from other software [but has been known to create duplicate entries in
  8. You can already do that, and have been able to for quite some time [since WDM drivers came out with Win98 if I'm not mistaken]. The choice is somewhere between the hands of the programmers and the users - between what the programmers want to stick you with [i.e. progs like SS that have no playback options at all] and what the user wants to use [i.e. foobar2000 and the like with ASIO, DirectSound, "WAVEOUT", kernel streaming, &c.]. I'll note that the DirectSound method [which allows individual control per program] has the added cost [sound-quality wise] of using a mixer that is either s
  9. SS's volume control uses the main control for everything ["Master Volume"] This affects everything coming out of your sound card.. Some programs use a control that is only for the stream they are playing [and won't affect anything else] through DirectSound. Others use the control for the WAVE output [which will affect anything playing back digital audio but nothing else]. Open your system mixer, play with the volume control in various programs, and you should be able to tell [immediately] which one they're using by which slider [if any] follows.
  10. Excellent response, tekdroid. My $0.02 is the a version: The RH1 is a recordist's unit and its interface as well as the additional features others are unlikely to use are optimised for recording use. Anyone who doesn't understand this after even a cursory glance at the unit with the display on deserves to get burned for not buying a player-oriented unit such as the RH10. "My $0.02 is the a version" should read, "My $0.02 is a short version"
  11. What exact format and bitrate [of aa3] were you using?
  12. ren *.aa3 *.oma There ya go. Also, side-issue: the atrac codecs for Sony's pro suite of tools won't open files with DRM on them [or at least, the version I have won't]. This means that any kind of transcoding [of, say, tracks you recorded yourself on your HiMD] must also be stripped of DRM before SF can even open them. I've been using SF since about 1992 or 93. It has its place, to be certain.
  13. Try http://users.pandora.be/satcp/eac-qs-en.htm As for the slight echo - what encoder were you using? I've been using EAC for years and have never had such a problem [not that I don't believe you are, of course].
  14. Doesn't EAC do this properly? I mean, I've used it for such with FLAC and MP3. I just don't recall if the tags were 100% right off the bat because I almost always mass-edit them with MP3BookHelper or Foobar2000 after encoding anyway.
  15. I missed that part. * Thou shalt not reply to forum posts after drinking a whole bottle of grenache-shiraz.
  16. If I'm not mistaken, a foot pedal was made for the [still being produced] "court reporter's" model of standard MD, which was made specifically for transcription purposes. Unfortunately I don't know the model # off the top of my head, but I do believe that of the 2 models I recall being like this, both were silver, fairly large even for MD, and had a built-in speaker. One also had a 2nd MD recorder in it specifically for duplicating/2nd-copying [for legal purposes] recordings as they were being made [this model I believe is NOT made any more]. In any case, Avrin is correct in that a foot p
  17. #1 - Why on earth are you using high sens mode to record anything louder than, say, a lecturer all the way across a hall, or birdsong? #2 - The way the gain on the mic input works appears to be like this: input -> preamp w/gain setting [low or high, ~15 and 35dB I believe] -> manual level control [this is NOT variable gain for the preamp, it's a level control AFTER the gain is applied, hence the preamp clipping regardless of what levels are set at - though in your case, turn off high sens mode, that should at least help a bit]
  18. Check out "Microphone University" at http://www.dpamicrophones.com/ . There are suggestions there on stereo techniques. I'd say picking a specific technique and sticking to it, and subsequently mastering the recording on a decent pair of monitors [they be speakers] - EQing, &c. - would go a long way. Expecting to get a perfect "sounds great on every system" recording without any editing, EQ, &c. is almost unreasonable. Still, I think the best advice of all is to master for the medium - listen to your recording on speakers when mastering, and take it elsewhere to check it out on o
  19. What used to be the case was that WAV files of artibtrary length [i.e. not an exact length of frames] would have trackmarks moved to the nearest whole frame boundary [probably back rather than forwrad, too]. WAV files of exact lengths in frames [75fps, 588 stereo samples/frame] would end up gapless with track boundaries where they should be. It appears that this may have changed [for the worse] around SS 3.4 to nearly random behaviour, though it does seem to make a difference what exact source the tracks are coming from [CD, CD image, WAV files, &c.].
  20. Yes, you're using your RH1 as an external DAC. There ar simpler ways [that are less stressful on the equipment] to do this though. Your turtle beach sound adapter likely already has a good output on it for headphones. If it lacks the power to drive 'phones, you'd likely be better off just using an external headphone amp. The write laser + magnetic head [since both are used for writing] should be in standby mode [the unit buffers data to be written, so it doesn't have to been written as it's converted in any case], but you're still causing wear to the equipment by leaving it in pause.
  21. Yes, MP3 on nearly all hardware players has gaps. ATRAC and its variants are among the only gapless lossy formats for portables on the entire consumer market. That's assuming you're dealing with a 1st-generation encoding [i.e. ripped directly from CD]. As to whether the MP3 bug is worse than the added artefacting of transcoding - that's up to the user's ears. The only way to find out is to try it. It also depends on what you're trying to encode, as different kinds of music [sound, rather] are more difficult to encode than others. I personally find that MP3s encoded at bitrates >192kb
  22. Despite what I've said about uploading post-editing [on the unit] working fine, I'd upload the entire thing -before- editing on the unit just in case. I'd still recommend as the least stressful and time-consuming methods either combining everything with SS [and adding trackmarks/splits in your editor after] or simply adding everything to the timeline of your editor and subtracting the bits you don't need. I'd get carpal tunnel trying to deal with that many trackmarks on the unit itself. For CD-burning, I tend to take my projects to a friend who has CD Architect. There are certain things
  23. --- Tag handling: After importing MP3s, since the "album view" of the library has no date fields in it at all, if a user sets the dates in this view [so that things will sort chronologically, which is part of how I catalogue all non-orchestral/classical/baroque/whateveryouwanttocallit music] then the DATE tag on some [but not all] of the tracks [i have been testing this with MP3s since SS2.1] will be overwritten with SS's proprietary, used by no other program I've yet seen date format [specifically, YYYY,MMDD or YYYY/MMDD depending on what you look at the tag with afterwards]. The files it s
  24. Notes I've made so far: Database usage is generally faster. This is good. Indigo is good. The interface still wastes way too much screen real-estate. Still no EQ. Consider: Winamp had an EQ under Windows95 on 32MB Pentiums. Other players have had EQs for years. In fact, OpenMG Jukebox and SS 1.5 had EQ. So really - what is the excuse not to have an EQ? Some of us use computer speakers, and without a bit of correction for said speakers or for room resonances, things are quite literally unlistenable. There are still issues with process priority. Certain operations [such an conversi
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