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portfair

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  1. Thanks for taking the time to respond. From the review which says "...all played gapless perfectly when I embedded 500x500px album art rather than the larger images I had previously embedded". As you've now confirmed it's about file size not image size.
  2. Hi Can you clarify this problem please? From reading the review it can be read that files over 500x500px present a problem but from reading the comments I think the issue relates to the size as in MB. I use a Linn ADS and the file I use for my 24/96 DSOTM is 1500x1500px at 176KB and the album plays gapless as expected. However if I embed a 6MB jpg in the flac files I do indeed get a gap during playback. Can you confirm if it is the size of the file in bytes and not pixels that caused the issue you experienced? If so I think this should be made clear in the review as this is an important issue for many people. I can think of no reason why an image file of MB size needs to be embedded in the flac files and for any reasonable use it would seem to me this is a non-issue. Thanks
  3. "Surely the housekeeping code can't be twice the data of the audio alone, can it?" Apparently (and I admit much to my surprise) it appears the answer is yes! From page 83 of The compact disc: a handbook of theory and use By Ken C. Pohlmann "A CD holding 74 minutes, 33 seconds of music contains 6.3 billion audio bits. However, after frame assembly and EFM coding, there are 19.3 billion bits to be stored. Thus only 32.65 percent of the disc holds audio bits. The rest is given to overhead” He then goes on to explain the overhead. I own an apology to the author at http://www.usna.edu/Users/math/wdj/reed-sol.htm, his maths seem to be spot on even though he doubts it himself.
  4. gordguide, In your post "Even today, getting a copy of the various books from Sony/Phillips....." you've written nothing that I disagree with or that I hadn't already seen elsewhere. But you don't address your statement that "a Redbook CD stores the data multiple times" and so far I've found nothing to support this. By the way I'm assuming that statement means what it says, that the audio data is stored at least twice. Is that what you meant to say? There just isn't enough space on a disc for this to happen. I also took at look at the link you provided and yes there is something seriously wrong with the authors maths! This seems a bit better http://www.usna.edu/Users/math/wdj/reed-sol.htm
  5. Thanks for responding gordguide I appreciate it. However I've searched and still cannot confirm that "a Redbook CD stores the data multiple times in multiple areas of the disk to provide robust error correction". It does delay and interleave data "thereby reducing the impact of burst errors" according to one site. In fact that's pretty much what it says on most sites that go into any technical detail. Could this be what you are referring to? Of course this isn't the same as storing the data multiple times.
  6. gordguide "you can be (almost) sure" - so in fact you can't be sure! "a Redbook CD stores the data multiple times in multiple areas of the disk to provide robust error correction" - that's not correct is it?
  7. “Submitted by The Computer Au... on Tue, 03/22/2011 – 00:26 ...The HDCD indicator on the Berkeley Audio Design Alpha DAC is a validation of bit perfection for normal playback of music...” I realise you're referring to data from a computer when you make this statement but I'm having problem making sense of it. If you use a CD transport as the source and the HDCD indicator on the Berkeley Audio Design Alpha DAC is on is it also an indication of a bit perfect data stream and that no error-correction has taken place while reading the disc? Surely HDCD discs are no more immune to data errors that standard red-book CD's. I would have though it much more likely that any corrected errors that are present are, by and large, of such a short duration as to have no effect on the HDCD indicator. If that is the case then it can't be a reliable indication of bit perfection from any source.
  8. On this very site there is a quote from Gordon Rankin from Wavelength Audio in which he says "Remember using these expensive cables on a hard drive is worthless they are in Block mode not Streaming and will not be effected by the use of costly cables". He was speaking of USB cables but I don't see why this wouldn't apply to firewire. See here Even on the DAC side you need to be careful. In this months Hi-Fi News (a British magazine just in case you don't know it) two USB cables are measured for jitter. The measurements are mentioned in an opinion piece and are not named but one is described as a 5 meter "bell wire" freebie the other a one meter highly priced audiophile cable. You probably know where I'm going with this but the 5 meter freebie had lower measured jitter than the audiophile cable.
  9. First off this is my first post and it's really long winded for which I apologise but I hope it may be of use to anyone who wants to extract DVD-V Hi-Res audio using Linux. I'm not sure how applicable it is to OS X but would be interested to hear if it works. I've also been grappling with ripping the Hi-Res audio from my Neil Young DVD-V's and wanted to try and do it without resorting to MS Windows which I'm using less and less. In this instance I'm using Linux (Fedora 12). Like the OP initially all I could get was 16 bit, 96 Khz using mplayer to extract the audio as a WAV file. I'm not sure but I think this might be mplayer reducing the number of bits to a level that my laptops on-board hardware can cope with. Anyway I've come up with this method which seems to work. The first two stages are at the command line but they are detailed below. The main applications/commands: lsdvd (command line) mplayer (command line) audacity (gui app) You will also need libdvdcss On Fedora these can be installed with: yum install lsdvd mplayer audacity libdvdcss Initially I had an error when using lsdvd but found a post that said something like “I always install all these applications and don't have the problem”. As it didn't say which application solves the problem I just installed the lot (actually some would have already been on my system but yum, the update mamager for Fedora will sort that out). So I ran the command from the post: yum install xmms xmms-mp3 xmms-faad2 xmms-pulse xmms-skins audacious audacious-plugins-freeworld* rhythmbox gstreamer-plugins-ugly gstreamer-plugins-bad gstreamer-ffmpeg amarok xine-lib-extras-freeworld mplayer mplayer-gui gecko-mediaplayer mencoder Note after loading the above I still got an error from lsdvd until after I had rebooted. Now to actually get the file off the DVD: First I run “lsdvd -a” and get the following: “libdvdread: Using libdvdcss version 1.2.10 for DVD access Disc Title: FILLMORE_EAST Title: 01, Length: 00:43:23.033 Chapters: 06, Cells: 06, Audio streams: 08, Subpictures: 32 Audio: 1, Language: en - English, Format: lpcm , Frequency: 48000, Quantization: 24bit, Channels: 2, AP: 0, Content: Undefined, Stream id: 0xa0 Audio: 2, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x81 Audio: 3, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x82 Audio: 4, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x83 Audio: 5, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x84 Audio: 6, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x85 Audio: 7, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x86 Audio: 8, Language: en - English, Format: ac3, Frequency: 48000, Quantization: drc, Channels: 6, AP: 0, Content: Undefined, Stream id: 0x87 Title: 02, Length: 00:00:06.166 Chapters: 01, Cells: 01, Audio streams: 01, Subpictures: 00 Audio: 1, Language: en - English, Format: lpcm , Frequency: 48000, Quantization: 16bit, Channels: 2, AP: 0, Content: Undefined, Stream id: 0xa0 Title: 03, Length: 00:11:37.043 Chapters: 09, Cells: 09, Audio streams: 01, Subpictures: 00 Audio: 1, Language: en - English, Format: lpcm , Frequency: 48000, Quantization: 16bit, Channels: 2, AP: 0, Content: Undefined, Stream id: 0xa0 Title: 04, Length: 00:11:37.043 Chapters: 09, Cells: 09, Audio streams: 01, Subpictures: 00 Audio: 1, Language: en - English, Format: lpcm , Frequency: 48000, Quantization: 16bit, Channels: 2, AP: 0, Content: Undefined, Stream id: 0xa0 Longest track: 01” Note that lsdvd reports the sampling rate incorrectly as 48000 even for the 96000 tracks. However it does identify Title 1 as the one I'm interested in as it's 24bit and has 6 chapters equating to the 6 audio tracks on Live at the Fillmore East, the DVD in question. Next I run mplayer: mplayer -dvd-device /dev/sr1 -vo null -vc null -ao pcm:nowaveheader:file="ny-1.pcm" -chapter 1-1 dvd://01 The options are: -dvd-device /dev/sr1 (This points to my DVD drive. If you only have one DVD drive this may not be necessary as mplayer should use it as the default but I find my laptop drive does not cope well with less than perfect discs so I also have an external drive I can turn to). -vo null (The video output driver. As I'm not interested in video set to null) -vc null (Video Codec – again not interested for this purpose so set to null) -ao pcm:nowaveheader:file="ny-1.pcm" (OK this I do want! As I said above extracting as a WAV didn't give me a 24 bit file so I extract as a raw PCM file, with no WAV header and output to a file named ny-1.pcm. -chapter 1-1 (Extract starting from chapter 1, ending at chapter 1. This exacts just the first chapter (track) into the file) dvd://01 (Tell mplayer it's reading a DVD and to start at title 1 as discovered using lsdvd). Run the command. When track one is extracted re-run the command changing ny-1 to ny-2 (that's just a new file name for track 2) and 1-1 to 2-2 to extract chapter 2. Keep going until all tracks are extracted. Should you want all 6 tracks in one file you can change 1-1 to 1-6 and run the command once (I assume - I've not actually tried this). mplayer will chuck out a bunch of text and then (hopefully) start extracting the data. Note one of the last lines mplayer displays is: AO: [pcm] 96000Hz 2ch s24be (3 bytes per sample) Note the following as they are needed for the next stage): 96000Hz (self explanatory) 2ch (again self explanatory) s24be (signed, 24 bit, big-endian) Once the trackes are extracted as raw PCM files I import them one at a time into Audacity, a free digital audio editor and save them as FLAC files. It is important that the import function is used as this allows you to specify the file parameters noted above. So run Audacity and then select: File -> Import -> Raw Data Navigate to the PCM you have created and open it. You should get a pop-up window with a number of settings. Set these based on those displayed by mplayer as noted above. So in this case set: Encoding to “Signed 24 bit PCM” Byte order “Big-endian” Channels “2 Channels (Stereo)” Start offset – leave as 0 bytes Amount to import - leave as 100 % Sample rate “96000” Hz Click “Import” The file should import. If you want you can test it by clicking the play button. I suggest keeping the volume down low just in case something has gone wrong and you get a horrible noise! To save as FLAC: File -> Export... A metadata pop-up may appear (this depends on the setting in audacity preferences). I generally leave it blank (I add tags later using EasyTAG) and just click “OK”. The next page is the save page. There is a drop-down menu in the bottom right which I set to “FLAC”. Then click the options bar at the bottom and set the “Bit depth” to 24. Click “OK” and then “Save” That should be it. If you want you can re-read the FLAC file back into Audacity and run a Frequency Analysis. File -> Open (no need to import the FLAC file) Select small amount of the file waveform using the mouse, then Analyze -> Plot Spectrum You may get a message along the lines of “Too much audio was selected”. Just click “OK”. The spectrum displayed should show frequencies up to 30 or 40 KHz. If anybody tries this I'd be interested to hear how you get on. If any body has an easier method to do this on Linux that would be even better.
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