raikis

I am trying to bookmark some of my analog transfers (RH1->3.5mm audio cable->audio recorder). For this I need to know exactly when each track starts and ends. I have found that it should be 86 frames within one second, but that does not look like it is the case. I have tried NetMDPython and it looks like it has around 85 frames per second, but not precisely. You can try adding up frames and checking it yourself (1st example: (80 + 84) - 85 = 79, 2nd example: 382 - 85 x 4 != 43). Maybe there is a bug in the script? I would love to try https://github.com/glaubitz/linux-minidisc/tree/master/libnetmd, but I cannot build it under Windows. Can anyone please explain how exactly should I interpret frames in NetMDPython? For example how to translate 85 or 50 frames into milliseconds? Why do I get neither 85 nor 86 frames per second when I have many tracks within one MiniDisc (the more tracks there is, the bigger the inconsistency)? Time used: 00:00:19+079 (0.26%) 2 tracks 000: 00:00:09+080 lp4 stereo protected 11 001: 00:00:09+084 lp4 stereo protected 22 Time used: 00:01:21+043 (1.01%) 8 tracks 000: 00:00:09+080 lp4 stereo protected 11 001: 00:00:09+084 lp4 stereo protected 22 002: 00:00:09+084 lp4 stereo protected 33 003: 00:00:10+010 lp4 stereo protected 44 004: 00:00:10+018 lp4 stereo protected 55 005: 00:00:10+022 lp4 stereo protected 66 006: 00:00:10+034 lp4 stereo protected 77 007: 00:00:10+050 lp4 stereo protected 88

I am looking for a way to get two spare parts for my MZ-RH1. My local spare part suppliers offer some parts for sale but not these two (look at the image attached). They say those parts are not stocked. Does anyone have any suggestions how to get them? Maybe there are similar parts that I could purchase (if so, what are their part numbers)?

Yes, this file is one of the larger ones I have transferred. However, I have managed to successfully upload 4 hour record (LP4). And yes, it took forever to complete and estimated remaining time and percentage figures were going crazy during that process. Yes, I have tried several times to upload that file and always got the same result. VLC can play that file fine before I run FCT and after I run it. When I transcode that oma file (before and after I run FCT) to wav using ffmpeg I get one of the two errors: Frame decoding error! Error while decoding stream #0:0 or JS mono Sound Unit id != 3. Frame decoding error! Error while decoding stream #0:0. So, based on all these finding I think that SS messes thing up and does not inform about any problem the end user. BTW, I use FCT from within SS all the time. File size of that oma file is 28.4 MB. And I am not sure I understand your question about fragmented cluster.

I have started my transfer process, but I have run into quite an unpleasant issue. By pure accident I have opened one of the transferred LP4 files in Sound Forge 11 and realized that Sound Forge could not read the end of the file, i. e. it was empty at the end, and I know for sure that there is audio data up to the last second of that particular track. To verify it, I have opened that oma file in VLC and it was playing fine from start to end. Then I have tried transcoding oma file to wav using SS, it did that, but instead of getting 60 minutes of record I got 55 minutes, so basically the last 5 minutes were chopped off. The biggest problem I see here is that SS did not report any errors while transferring track or transcoding it. So, in theory I need to check every single uploaded oma file to be sure it is not blank or otherwise damaged. Can anyone suggest any better approach as it will be very time consuming to check hundreds of oma files manually?

I have just purchased a MZ-RH1 in order to transfer my old records, that were done using an external mic and Net MD device, to PC. I am looking for the best way to archive those records. I have a few questions: 1. Since all my recordings are done in standard MD mode (mostly LP4), I won't be able to use HIMDRenderer, QHiMDTransfer or any other program to transfer my records. Only SS will work, right? 2. SS allows to transcode transferred *.oma files to *.wav files and I should use this feature to get DRM-free audio files, right? 3. There is no way to losslessly transcode *.oma files to any other format without increasing file size massively, right? For example, 10 Mb *.oma produces 210 Mb *.wav file and if I compress it to *.flac I still get 78 Mb - that's almost 8 times larger file. In other words, there are no tricks to further losslessly compress and get smaller file size, right? 4. *.oma files created by SS still have DRM protection even though it lists the following info under file's Properties? Playback restrictions: None No. of times played: 0 times No. of times transferred: 0 times Remaining transfer count: Unlimited Remaining ATRAC CD transfer count: Unlimited Remaining audio CD transfer count: Unlimited 5. To get rid of DRM protection I asked in (4) I have to use File Conversion Tool. This will assure that no DRM protection is left within my transferred *.oma files, right? 6. If I have used SS to transfer not protected music to my Net MD device, even with MZ-RH1, I will not be able to upload it back to PC, right?