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Found 11 results

  1. For music and movie streaming, Im using Amazon Prime normally. Since longer Im searching for possibilities to extract mp3-files out of this streaming plattform, that I don`t have to rely on the application always, expacially in order to listen to music via devices without internet connection. Anybody knows, if there is a legal way to do so? Do you have some recommendations regarding to specific software programs? Thanks for advice in advance!
  2. I am concerned that Apple's pushing of Apple music in the HomePod presentation is a sign of worse days ahead for people who store music files locally. Politically speaking, the best way to lobby against this trend at Apple might be to speak on behalf of people with cabins or rural homes that have power but no internet. Or on behalf of the African, Latin American and other cultures whose music from the past 90 years is only sparsely catalogued by Apple Music. Why can't Apple (or someone) come up with a play-only head unit that will run iTunes, iPod style, from (a clone of) an external drive containing the user's library and music files, and be able to control devices like the HomePod via Bluetooth, Lightning or USB 3.1 or C—but using a bare-bones play-only version of iTunes, not iOS Music. The iOS Music app seems hobbled in so many ways, presumably to force people into using web-dependent Apple Music. Anyone with rare, exotic or out-of-fashion music is stuck with Apple Music's boneheaded track and artwork mis-identifications.
  3. Note: This is going to be a little complicated, so I will try to cover every factor in this problem I've had. I purchased some DSD (352 khz) files of solo piano music that's fairly dynamic and detailed, with loads of gorgeous harmonics. I prefer to play my music in 24/88 or 24/96 format, and also be able to convert to 16/44 WAV to play on my 256 gb Apple devices. So I purchased a well-respected converter (Mac version) to make the 24/88 FLAC and WAV files, and carefully followed the instructions given by the music critic who promoted the album, as well as the detailed instructions given by the converter gurus. Those 24/88 conversions not only wouldn't play on my PC in Foobar2000, they corrupted Foobar so badly that I had to reinstall it. That process of conversion from DSD to 24/88 and attempts to play on Foobar followed by reinstallation was repeated many times, with input from the persons noted above, with no success. Note that I have over a thousand high-res (24/88 to 24/192) WAV and FLAC files from HDTracks and several other such sites, and I've never had a single problem playing those on my Foobar2000 player. And I tested a number of those again during this process. So I finally decided to convert the DSDs to 16/44 WAV files, and those played just fine on Foobar2000. But when I converted those WAV files to FLAC, the FLAC files (Level 5 FLAC) were smaller than MP3s, i.e. about 23 percent of the 16/44 WAV size. Using that same Foobar2000 and both FLAC converters I have (ca. 2007 and 2014), I've converted several thousand CD/WAV rips to FLAC over the years, and none of those FLACs were less than 60 percent of the WAV file size. I even went a step further to see how my trusty FLAC converter could make files smaller than the corresponding MP3s - I converted a few MP3s (320 kbps, 4 minutes and 10 mb each) to 16/44 WAV tracks (44 mb each), and then I restarted Foobar and converted those WAVs to FLAC. Those FLACs, which originated from the MP3 files, were now 24 MB each, and Foobar showed a bitrate around 760 kbps when playing. The "23 percent" FLACs noted above showed a bitrate playing in Foobar of 290 kbps. So my question is, given that I cannot make a FLAC that's 23 percent of a WAV file size from any of the thousands of WAV files that I've ripped or purchased, or even from up-rez'd MP3s, what could possibly be in the WAV files I got from the well-regarded Mac converter that I purchased, that would trick my FLAC converter into making FLACs smaller than MP3s? The converter software gurus say that the DSD recordings have "extremely low complexity", which makes no sense, and doesn't fit with all of the other experience I've described.
  4. Seems everyone has very specific, even finely tuned preferences as to how they enjoy their music listening setups. Now with digital we have so many more options! I found it somewhat overwhelming in the beginning, and still find parts of the system confusing and confounding! I wonder if others are as curious as I with regards to how YOU listen to YOUR Digital setup. Preferred format: DSD, FLAC, ALAC, .Mp3... Player/Renderer of choice? Which Server software/hardware? Asset, JRiver, Foobar, Kazoo, etc. Storage on NAS, dedicated hardware.....or? Do you like/use a streaming service? Which one(s)? How do you organize/sort your way to play tunes? By Folder, Album, Artist, Jukebox, ..etc Biggest issue/problem currently, or in the past... Thanks for sharing! If this has already been done, well, I couldn't find any threads... looking forward to reading about you all! BigBadger
  5. https://www.npr.org/sections/therecord/2015/06/02/411473508/how-well-can-you-hear-audio-quality 6 tracks. 3 encodings 128/320 kbps MP3 and 16/44.1 Take it as many times as you would like until you are perfect at it :-)
  6. Like lots of people, I have several hundred CDs that contain music I really like. Pandora is great, but not as good as my CDs. Recently, my old Sony ES amplifier died after 30 years, and I bought a Sony STR-DN1030 to replace it. My problem is to figure out how to be able to play the CDs and playlists based on the CDs using reasonably friendly software. I know I can get the CDs ripped and converted to mp3's, and then put on an external HD or a USB flash drive. But after that, I am at a loss. The new AV receiver supports connecting to my iPhone via Bluetooth and Apple AirPlay, but the software is terrible. The new AVR also supports connecting to my notebook computers (via wireless or wire) and I can play music in folders or in playlists. But, again the GUI on the TV is just terrible. I can't read the titles or artists, and the song durations and playlist contents are not shown. The GUI that is so terrible seems to be part of the software that comes with the Sony STR-DN1030 AV receiver. It is just unacceptable. If I had an extra computer with an HDMI port, I could connect it to the receiver, open up iTunes or Window Media Player, and push what it was putting out into the receiver, which would then send the video to the TV and the sound to the speakers. I suppose if I got a wireless mouse and keyboard, I could control it all remotely, i.e. from the couch or dining room. Is there simple (cheap) hardware dedicated for this purpose? It just needs an OS, a hard drive, and HDMI port, and the ability to run music playback software. Any advice to this newbie is much appreciated.
  7. In the past I have been able to convert entire albums to MP3 files so that I could load them onto a Sandisk MP3 Player. but when I did this, all of the albums ended up with each song duplicated;..one song as WAV and also as MP3. I just created an playlist (300+ songs) from multiple albums. I want to convert the entire playlist to MP3 files so that I can load them on to another MP3 Player. As a test, I did convert 1 song from the playlist to MP3 and the process appeared to happen sucessfully; but when I view the file type of that song within the playlist, it still shows up as a WAV file. So my question is; Can I convert the entire playlist (from playlist) to MP3 and have it as a new MP3 playlist? Any insight is apprectiated!
  8. Hi-res Wav vs Flac vs CD vs Mp3 free comparison files Downloads: Link to Zip File ; Compare Formats
  9. Maybe it's because of the business I'm in, but I define compression as fitting everything that is there into a smaller space. Zip files, Flac files etc., or as we call it lossless. MP3's and the like conflate data. They, arguably, with intelligence, throw out data to make the file smaller. So why aren't MP3 referred to as conflated ? Arguably, I could be considered one with little intelligence writing about such a trivial peve! Larry
  10. I spend in car 1-2 hours everyday. In car I listen CD-audio disks only. As qualitative source. I'm interested who use like and other sources in your cars? Tell about your equipment (head device only, iPod, iPad etc.), prefferable formats, why you listen this formats?
  11. I was very intrigued by an earlier post on this forum where it was suggested that Bass lines would be particularly sensitive to the effects of compression using the pyscho-acoustic algorithms employed in MP3 encoding. Presumably this is because low levels of low frequency information are easily ‘masked’ by louder, higher frequency information in the material and thus are removed in the compression. I thought it might be interesting to take a look at that idea ..... So, simply put, from the diagram above, against the backdrop of an equal loudness plot of the human ear’s sensitivity, you can see the depiction of a louder sound ‘masking’ another sound. In these circumstances the louder sound obscures the quieter sound and in the compression of this sequence the quieter material will not be encoded and, thus, it’s ‘lost’ from the final compressed material – hence the term ‘lossy compression’, of which MP3 is one variant. Coming to the specific case of the bass instruments, I was keen to establish how its sound in particular might be impoverished in such compression. I guess the first thing to consider would be the frequencies involved and also how the harmonics of bass notes might be affected by the MP3 compression algorithms. In the diagram above you can see the fundamental note in the middle of a bass’s musical range and the notion of its quieter harmonics superimposed on an equal loudness curve. It’s important to remind ourselves here that an instrument’s ‘recognisability’ is a function of its musical range, its dynamics and in particular, its timbre. Timbre is what allows us to distinguish two instruments playing a note at the same pitch and loudness and it is due to the strength of each instrument’s production of harmonic overtones. All other things being equal, different harmonic overtone content results in different characteristic timbres. What we get from this, is that if you significantly alter the representation of a note’s harmonic content, it is very unlikely that you will faithfully represent the full timbre of the note and therefore the timbre of the particular instrument producing it. Of course, the fundamental frequency of the note dominates in loudness terms, so its frequency will give the game away in the case of a bass note, but it is quite possible that the full character of the bass note will not be rendered accurately if much of the harmonic content of the overall note is ‘masked’ out. In a full mix of instruments playing simultaneously, the acoustic or musical ‘space’ that a bass note’s distinguishing harmonic overtones occupy would fall in the mid-frequency range where that space is very likely occupied by louder fundamentals of other instruments’ notes. In this situation, it would be very likely that the bass note’s harmonic content would be ‘lost’. It’s interesting to look at a spectrum analysis of a double bass’s note – both bowed and plucked to appreciate the harmonic content. In the analysis below you can see that the harmonics, in the case of the bowed G note, are a very significant part of the note’s overall frequency content, which rises to well over 5Khz, and you can also see that they persist for quite some time after the leading edge of the note is passed. Plucking the bass note would appear to significantly damp the production of harmonic content above 1Khz and is likely due to the fact that the bow physically excites the string’s vibration for longer, allowing more harmonics to develop. Looking further into the harmonic content of bass notes, I analysed a track that has a lot of individual bass notes played in isolation and also passages where the same bass is part of a complex mix. The spectrum analysis above shows a single plucked double bass note from the track ‘Bass Suite No1’ from Avishai Cohen’s 1998 ‘Adama’ album. The left hand plot is from an MP3 CBR 128Kb/sec encoded by the LAME 3.98.4 encoder; the right hand plot is from the 16bit / 44.1 Khz WAV file, You can see as before the MP3’s low pass filter cut off at just above 16Khz. You can also see the relatively strong harmonic content of the single note in both the MP3 and the WAV plot. The WAV plot. on the right, shows harmonic content all the way up to 20Khz (the top of the spectrum stops at 22Khz). The amplitude of the note’s fundamental, at the very foot of the plot, has been over-scaled to allow the plot to illustrate the harmonic content. So, you can see that the harmonic content is significant. If we put these two traces together, MP3 and WAV, you see more readily see the comparison:- Now let’s look at a sliding single bass note. I have isolated this note in the time domain so we can easily see it’s spectral analysis in the mix:- Here, I have highlighted, in white, the fundamental and the first 15 Harmonics of the bass note as it slides up in frequency. You can see there is again significant harmonic content – in this case clearly visible all the way to nearly 2Khz. It’s slightly easier to see if I extract this note and it’s harmonic content into its own spectral layer and show the relative strength of the harmonics in 3D so you can see how each harmonic varies in amplitude over the life of the sliding bass note. You can also see the ‘bump’ of the fundamental at the bottom left of the plot:- Now let’s look at three notes in quick succession. Here is a plot of three bass notes plucked in quick succession, which have a sound like ‘DAAH-dit-Dummm’:- You can see again, in the WAV plot, that there is a strong fundamental in the first note. Less in the second, and almost as much in the third note as the first, but, there is also very strong harmonic content accompanying each note in the whole phrase. Just how strong these harmonics typically are can be seen in the next plot:- Considering the fundamental peaks at -22db and even the 10th Harmonic is still at -60db, it shows, since harmonics reduce in loudness as they increase in frequency, that the lower harmonics would have significant influence on the ‘character’ i.e the Timbre of the overall bass note and therefore preserving them would be essential when the bass notes are being ‘solo-ed’ as in this case. In the combined plot below you can see that the MP3 version of these three notes, apart from the content above 16Khz which has been removed by the encoders low pass filter, is pretty much identical to the original waveform in the WAV file:- In the combined plot above the green trace is the original WAV file content of the three notes and the slightly offset yellowish coloured trace is that of the compressed MP3 representation of the same notes. So, in the case of ‘solo-ed’ bass notes MP3 compression does a pretty good job. But what would happen if the bass notes were being played in the middle of a complex musical passage? This Adama track includes contributions played on the double bass, a soprano saxophone and a trombone, together with percussion instruments. Leaving out the percussion instruments, on the basis that their notes are more like impulse notes and of relatively short duration, we have a musical frequency ‘space’ which looks like this:- From this chart you can see that continuous notes played on the saxophone and the trombone would occupy the same frequency ‘space’ in the mix as the principal (loudest) harmonics of a typical, relatively short duration double bass note and therefore might very well mask those harmonics so that they would not be heard. In that case you might expect the MP3 compression’s psycho-acoustic algorithms to ‘lose’ the bass harmonics. The next spectral analysis shows a passage where, at the beginning of the time window, there is a bass note played pretty much in clear ‘space, and at the end of the window you can see another bass note played entirely on its own:- In the red highlighted box though, you can see a bass note that has been played at the same time as both wind instruments are playing a longer, louder phrase. In the red box you can just about also see the trace of the wind instruments notes’ true amplitude raised as a single line above the 3D spectral trace (the spectral trace of the wind instruments’ fundamentals and lower harmonics are deliberately clipping in this plot so that we can more easily see the overall content, which is why they appear to have flat tops in the lower frequencies). As before, the green traces are of the WAV file notes and the red show the slightly offset MP3 representations of the same notes immediately alongside. In the white box you can see, particularly in the centre of the image, how it appears that the bass’s harmonics are partially occluded by the wind instruments in the WAV trace but very significantly affected in the MP3 representation. It is difficult to see any of the bass’s harmonics represented at all in the MP3 companion trace in this more complex passage, whereas, in the areas where the Bass notes are on their own in the mix, the MP3 representation clearly matches the WAV file – see the note at the end of the time window. It’s perhaps easier to see the WAV / MP3 differences in this centre part of the passage where you can see multiple differences in the representations:- So, all in all, an interesting experiment which shows that where instruments are soloing or are the predominant sound in the mix, the MP3 algorithms do a pretty good job of representing fundamentals and principal harmonics. Where the mix is much more complex, however, the picture is not so clear, and the representation seemingly not so faithful to the original waveform. But that, in a nutshell, is precisely how ‘lossy’ compression works after all – it does not encode what the ear and brain will supposedly, according to the theory, not ‘hear’.
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