I think db1989 understood what you meant and suggested how to change the sample rate in the third post to this thread.
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192kHz FLAC
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Best YouTube's audio
QUOTE (sintapilgo @ Aug 13 2013, 04:55) <{POST_SNAPBACK}>
So, although the higher bitrate of the new file, I think the old file have better quality because it is more original/less modified".
Am I right?
Am I right?
Did you try a ABX test?
Also, a spectrum analysis might reveal a few things without the need for a BX test.
If I where to take a wild guess I'd say that Youtube (for both audio and video) uses a intermediary format to store a "master", this may or may not be lossy or very high bitrate lossless.
It is also possible that Youtube do keep the original uploads if they are within certain desired parameters of their automation system. Search the net and see if you find any info about their process.
One thing I feel somewhat confident about is that Youtube most likely does not upconvert anything. I.e. going from low bitrate to a higher would be silly and potentially degrade the audio.
It would also be a bad utilization of bandwidth.
Now this is original uploads we're talking about. When it comes to mashups etc. then some people use youtube videos as their sources and then upload to youtube again, what's happen to the video or audio at that point is anyone's guess, but depending on the lossy formats bitrates it wont take many rounds of that before you start hearing artifacts.
I did some uploads where the audio was lossless (uncompressed PCM), my guess is that those are stored at youtube's servers someplace as FLAC or (something else lossless).
Youtube should provide you with the best quality it can at the bitrate it gives you. So 360p is more lossy than a 1080p stream for example. (the audio follows suit as well).
I have seen some Streams that never are above 360p or 480p, it is possible the original upload was never better than that, hence no 720p for example. I'm guessing the same is true with the audio in that regard.
I.e: Garbage In = Garbage Out
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Metadata/tag mapping between FLAC, mp3, and UPnP/DLNA
QUOTE (Rescator @ Aug 13 2013, 16:06) <{POST_SNAPBACK}>
You might want to re-think that and leave it blank instead.
Why? Because some software might treat 0000 as 0 or 00 and numerically speaking could interpret that as being 2000
While I can't recall seeing two digit years, its' possible, as it is just a text field after all.
At least with blank you can still sort them, and it would reveal odd behaving software that do not handle empty year fields.
Also note that I've seen full dates in the year field as well. In the wild I think I've even seen one with the time.![blink.gif]()
I guess if the field length in the tagging/file formats out there allow it then the year field could be "changed" to accept a ISO 8601 date (see http://en.wikipedia.org/wiki/ISO_8601 )
And if no timezone is indicated then Zulu should be assumed.
THe current year only field would be somewhat compatible, the issue though is dates. And I have seen in the wild that / has been used which is not supported by ISO 8601. And then there is the international vs US way of month-day vs day-month ordering.
If anything is ever done to standardize the year/date field then my suggestion is make it ISO 8601 compliant.
Why? Because some software might treat 0000 as 0 or 00 and numerically speaking could interpret that as being 2000
While I can't recall seeing two digit years, its' possible, as it is just a text field after all.
At least with blank you can still sort them, and it would reveal odd behaving software that do not handle empty year fields.
Also note that I've seen full dates in the year field as well. In the wild I think I've even seen one with the time.
I guess if the field length in the tagging/file formats out there allow it then the year field could be "changed" to accept a ISO 8601 date (see http://en.wikipedia.org/wiki/ISO_8601 )
And if no timezone is indicated then Zulu should be assumed.
THe current year only field would be somewhat compatible, the issue though is dates. And I have seen in the wild that / has been used which is not supported by ISO 8601. And then there is the international vs US way of month-day vs day-month ordering.
If anything is ever done to standardize the year/date field then my suggestion is make it ISO 8601 compliant.
You might be right about leaving the dates blank. I was only thinking of using "0000" as the software I use (MediaPortal) didn't like blank Album or Album Artist fields so I wasn't sure if it could handle blank dates properly either. I guess I should check for sure before I decide what to do.
I've seen full dates in there as well but personally I don't see the need for anything more than the year of release.
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Best YouTube's audio
QUOTE (saratoga @ Aug 13 2013, 06:43) <{POST_SNAPBACK}>
But why are you assuming they didn't just create all the transcodes from the source when it was uploaded?
From the audio-only file: "Encoded date: UTC 2013-07-21 16:59:07"
The upload was done years ago...
QUOTE (saratoga @ Aug 13 2013, 23:19) <{POST_SNAPBACK}>
I think theory is pointless here unless you can find out for sure what youtube does. If you're just guessing how they transcode, theory is only as good as your guess.
Maybe someone could recommend a software that analyze if the 254 Kbps is a compression or bloated from smaller bitrate.
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HE-AAC gapless playback
Extra add for this, the XLD encoder plug-in based on fdk-aac has the same problem with SBR. I found that subtracting 481 from the delay written to SBR files worked perfectly, but not so for SBR+PS. It seems SBR+PS need to subtract 391 from the combined delay given by the encoder instead.
EDIT: Scratch that, I was using Jack rerouted output instead of actually converting the files. Seems like 481 is correct for PS as well.
EDIT: Scratch that, I was using Jack rerouted output instead of actually converting the files. Seems like 481 is correct for PS as well.
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Displays FLAC embedded full size album art
QUOTE (99Ehsu @ Jul 2 2013, 15:22) <{POST_SNAPBACK}>
QUOTE (Tim De Baets @ Jun 30 2013, 04:02) <{POST_SNAPBACK}>
I have just tried it myself and indeed, the PlayerModeAlbumArtSelected and ShowAlbumArt settings don't seem to have any effect on the album art of FLAC files (and probably not on the album art of any of the other non-native formats either).
For the moment, I don't know why this could be happening so I'll need to investigate it further. I'll do so when I have some time and if it turns out to be something that can be easily fixed, I'll include the fix in one of the next versions of WMP Tag Plus.
For the moment, I don't know why this could be happening so I'll need to investigate it further. I'll do so when I have some time and if it turns out to be something that can be easily fixed, I'll include the fix in one of the next versions of WMP Tag Plus.
Thanks for your effort!
Any progress or finding to display the larger Art Album works?
Waiting for your good news!
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Now that you found your lowest transparent setting...
Archival: 3 independently lossless ripped libraries of music I really love on separate hard drives; FLAC, ALAC, and WMAL. Balance of "so-so" music is on the largest drive ripped lossless with any of the aforementioned 3 I felt like using at the time.
Home/Desktop: A mix of lossless (ALAC), aac (either iTunes plus or equivalent qaac setting of 256kbps cvbr), lame V0, halb27's lame V0 cvbr0 "insane" setting, and a few misc. purchased tracks/albums from various sources (mainly GooglePlay or iTunes).
Portable Flash Drives (for work): Mainly lame V0 for compatibility with any equipment.
Rockbox'd E280R 8gb with 32gb micro sd card: Mostly Vorbis aoTuVb6.03 at the .q5 quality setting. Some misc. aac and mp3 files.
Rockbox'd Sansa ClipZip 8gb with 32gb micro sd card: Almost all FLAC level 4 with a few purchased aac and mp3 favorites.
My wife's Rockbox'd Sansa ClipZip 4gb with 32gb micro sd card: Mostly Vorbis aoTuVb6.03 at .q6 and .q5 quality settings. Some misc. aac and mp3 files.
Kid's Rockbox'd Sansa Clip 2gb: Mostly Vorbis aoTuVb6.03 at .q4 quality setting.
No transcodes on anything, ever. If I can't purchase in cd or lossless, the original purchased file stays "as is" except for added metadata or ReplayGain or SoundCheck info.
Home/Desktop: A mix of lossless (ALAC), aac (either iTunes plus or equivalent qaac setting of 256kbps cvbr), lame V0, halb27's lame V0 cvbr0 "insane" setting, and a few misc. purchased tracks/albums from various sources (mainly GooglePlay or iTunes).
Portable Flash Drives (for work): Mainly lame V0 for compatibility with any equipment.
Rockbox'd E280R 8gb with 32gb micro sd card: Mostly Vorbis aoTuVb6.03 at the .q5 quality setting. Some misc. aac and mp3 files.
Rockbox'd Sansa ClipZip 8gb with 32gb micro sd card: Almost all FLAC level 4 with a few purchased aac and mp3 favorites.
My wife's Rockbox'd Sansa ClipZip 4gb with 32gb micro sd card: Mostly Vorbis aoTuVb6.03 at .q6 and .q5 quality settings. Some misc. aac and mp3 files.
Kid's Rockbox'd Sansa Clip 2gb: Mostly Vorbis aoTuVb6.03 at .q4 quality setting.
No transcodes on anything, ever. If I can't purchase in cd or lossless, the original purchased file stays "as is" except for added metadata or ReplayGain or SoundCheck info.
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How to convert 192 kHz FLAC to lower sampling rate with fb2k or other?
QUOTE (NuKleos @ Aug 14 2013, 00:26) <{POST_SNAPBACK}>
I tried SoX 14.4.1 but I couldn't get it to run after installation,
What specifically did you do with sox? Your requirement is a matter of seconds with sox
CODE
sox input-file.flac -r 44100 output-file.flac
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Three SPDIF outputs, but only two inputs
A single device with two independent spdif inputs is rather complicated. SPDIF carries its own clock and two spdif receivers generate independent clocks, i.e. incoming streams with independent timing which are not simple to handle in a single interface. If your SPDIF signals originate from a single clock domain (master clock, single soundcard with multiple spdif outputs (e.g. Audigy 2 with 4 spdif pin headers), it makes it much easier.
Or just get two independent soundcards with single spdif inputs, but you have to make sure you do not need to process their inputs synchronously (i.e. multichannel or merging them in a single dsp processing chain).
Or just get two independent soundcards with single spdif inputs, but you have to make sure you do not need to process their inputs synchronously (i.e. multichannel or merging them in a single dsp processing chain).
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Has anybody ABX'd headphone/speaker burnin?
No doubt, they will all improve after break-in regardless of the variation whether it exists before or after burn-in.
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CD Accurately Ripped in foobar, but not in CUETools?
Thanks, for answering rapid and qualified ![smile.gif]()
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Normalizing very old music
QUOTE (saratoga @ Aug 13 2013, 22:07) <{POST_SNAPBACK}>
Burning PCM to a CD is a lossless operation
Only if the PCM is 44.1kHz 16-bit, and only if you have "normalise audio" unchecked. Some old versions of Nero had that checked (on!) by default, so all tracks get peak normalised - shouldn't clip though. Some old CD burning software would upsample lower sampling rates (e.g. 24kHz) to 44.1kHz simply by repeating samples - that adds horrendous metallic high frequencies.I doubt either is the problem here. Just saying.
QUOTE (AndyH-ha @ Aug 14 2013, 07:15) <{POST_SNAPBACK}>
not new technology
I'm liking the re-assurance! Cheers,
David.
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MP3 repacker
Latest version 2.04 is much faster than 2.03 at running --ib but occasionally locks up. Trying this on a group of mp3s will cause it to lock up on different files. 2.03 does not lock up on the same set of mp3s. Happens with 32bit and 64bit versions.
** correction, 2.03 locks up as well. Also, doing it repeatedly on any random mp3 will eventually cause it to lock up.
Even more interesting, I got the 64bit version of 2.03 to lock up when displaying the help info (by giving no input file). It showed everything down to --help and then froze.
** correction, 2.03 locks up as well. Also, doing it repeatedly on any random mp3 will eventually cause it to lock up.
Even more interesting, I got the 64bit version of 2.03 to lock up when displaying the help info (by giving no input file). It showed everything down to --help and then froze.
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How to determine gapless information in runtime
QUOTE (Alex B @ Apr 4 2011, 06:39) <{POST_SNAPBACK}>
It would be nice if someone could explain how the Nero encoder uses the chapter list for storing the gapless decoding data.
As far as I know, older version of Nero encoder used to employ non standard Nero style chapter list (udta.chpl) to declare delay, and stts (Decoding Time to Sample Box) to denote valid length of the final frame.
If you don't know about MP4 stts box, you can roughly take stts as a table keeping length of each frame. For constantly framed audio codec like AAC, it usually has only one entry, whose length is 1024 or something.
In case of Nero, it has an extra entry for the final frame, which keeps valid length of the final frame (which is shorter than 1024, and a player has to trim decoded audio down to that length after decoding).
This is the same method as is used by ALAC in m4a.
Finally, since Nero style chapters can be used for ordinary purpose, I think there rises an ambiguity on treatment of the first entry in the chapter list.
It seems that fb2k always expects first entry to denote the delay when Nero style chapter is present.
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Personal approaches to tagging, part 9001
What I wrote applies only to iTunes on the computer, unfortunately iPods don't do the intelligent grouping. What I do in those cases is set the ITUNESCOMPILATION tag to 1, which makes the album appear only in the Compilations menu on the iPod and not the Artist->Album menu (unless you have other albums by that artist, in which case the artist-specific tracks appear under the respective artists).
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Denon mixer question- multiple digital inputs
QUOTE (Spikey @ Aug 14 2013, 08:41) <{POST_SNAPBACK}>
All the devices I'm using are 44.1 khz, none 48, as well. I'm just wondering whether this will work.
Thanks Arnold. Your posts are always helpful. I'll try it and post if there's anything insightful to be gained.
Thanks Arnold. Your posts are always helpful. I'll try it and post if there's anything insightful to be gained.
It may not be well known that you can use most hardware sample rate converters with approximately the same sample rate in and out. In that case, they correct for any clocking issues.
There are off-the-shelf high performance SRC chips that are not excessively costly.
Here's an example of a very high performance chip for the purpose:
http://www.analog.com/en/audiovideo-produc...ts/product.html
They are about $4 in single units, so maybe half that or less in production quantities.
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Surround sound mapping in Audacity
QUOTE (Maggi @ Aug 14 2013, 16:17) <{POST_SNAPBACK}>
sorry, I have no idea what's going wrong on your end ... ![sad.gif]()
Thanks for trying.
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Metadata/tag mapping between FLAC, mp3, and UPnP/DLNA
Here is my attempt. It is not finished yet! I intend to cull the items which don't map, but I'm leaving them ALL there for now to see if there's a soft mapping (TXXX or IPLS/TIPL) which works. I intend to organise it better too. Other formats can be added fairly easily from the URLs at the bottom of the table, but later. Anyway...
http://wiki.hydrogenaudio.org/index.php?title=Tag_Mapping
Here are some problems...
1. When inventing new frames for ID3v2, some sources just invent a new frame (especially a new ID3v2.3 frame with the same name as a real ID3v2.4 frame - but sometimes an entirely new T frame), some sources invent a new frame with an X at the start, and many use TXXX:framename (which is the correct method according to ID3.org). Any thoughts on these three approaches? Anyone seen yet another approach?
2. With Vorbis Comments, Who/what is using TOTALDISCS in preference to DISCTOTAL? ditto TOTALTRACKS vs TRACKTOTAL? If it's lots of things, then ideally you need to read+write both versions for compatibility, but the xiph.org docs only mention DISCTOTAL and TRACKTOTAL.
3. Given the now common mapping of TPE2 to Album Artist, is there a field left for Orchestra/Band? Given that TPE1 is mapped to Artist, it also loses its original meaning of "soloist". I guess soloists can now be listed first in one of the performer/musician/IPLS/TIPL fields, leaving TPE1 free for Orchestra? I only ask because I'm wondering how many IPLS/TIPL/TMCL vs Vorbis Comments mappings to figure out.
It gets worse if you assume that, when tagging classical music, some people will abuse Artist=composer - but that's for another day (and thread!).
Cheers,
David.
http://wiki.hydrogenaudio.org/index.php?title=Tag_Mapping
Here are some problems...
1. When inventing new frames for ID3v2, some sources just invent a new frame (especially a new ID3v2.3 frame with the same name as a real ID3v2.4 frame - but sometimes an entirely new T frame), some sources invent a new frame with an X at the start, and many use TXXX:framename (which is the correct method according to ID3.org). Any thoughts on these three approaches? Anyone seen yet another approach?
2. With Vorbis Comments, Who/what is using TOTALDISCS in preference to DISCTOTAL? ditto TOTALTRACKS vs TRACKTOTAL? If it's lots of things, then ideally you need to read+write both versions for compatibility, but the xiph.org docs only mention DISCTOTAL and TRACKTOTAL.
3. Given the now common mapping of TPE2 to Album Artist, is there a field left for Orchestra/Band? Given that TPE1 is mapped to Artist, it also loses its original meaning of "soloist". I guess soloists can now be listed first in one of the performer/musician/IPLS/TIPL fields, leaving TPE1 free for Orchestra? I only ask because I'm wondering how many IPLS/TIPL/TMCL vs Vorbis Comments mappings to figure out.
It gets worse if you assume that, when tagging classical music, some people will abuse Artist=composer - but that's for another day (and thread!).
Cheers,
David.
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WavPack on 64-bit OS?
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Closed Back Headphones like Senn HD600
Any of you guys follows Rins blog? http://rinchoi.blogspot.com/
Lately he's measured the HD650, HD280, K702 and publishes an additional graph on each headphone he tests conforming to the new Olive-Welti target, from the same people that published "The Relationship between Perception and Measurement of Headphone Sound Quality" a while back.
Lately he's measured the HD650, HD280, K702 and publishes an additional graph on each headphone he tests conforming to the new Olive-Welti target, from the same people that published "The Relationship between Perception and Measurement of Headphone Sound Quality" a while back.
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