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  • audio quality question

    I have noticed that most of my music and movies sound great but sometimes I will listen to an album or watch a movie and the audio is terrible. My speakers make crackling sounds as if they are blown during any midrange frequencies. I know its the quality of my download or rips but what do I need to look for so I know if it will have good sound. Is there a bitrate or are wma's better than mp3...etc. I just dont want crappy sounding audio in my library so I now need to go thru and delete the bad onesthat should take forever.
    Also, is there any software that you can run audio files thru that will "upconvert" them or clean them up? Sorry if its a stupid question but never really had to deal with this before.
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  • #2
    Itunes is pretty good. Bit rate means nothing in downloads. Idiots take a low bit rate, convert it to a high, and then upload it as if its actually good quality. Look at reviews of each "site" where you download music.
    Acer Netbook with 160hdd and 1.60 Ghtz atom - to be installed


    • #3
      Either that or rip music yourself into the format and bit rate to suit your audio quality wants/needs.

      If an audio file is already at a low bit rate, then upconverting it to a higher bit rate isn't going to improve the quality of that music file.
      128kbps is supposedly CD-quality. I don't get music files for my collection that are lowe than this.
      Many music sites have 96kbps files or even lower quality. It's a shame, but you have to consider that there are a lot of people out there who can't tell the difference.
      I don't get that because I have some hearing loss in one ear and I can tell the difference between a 128kbps MP3 file and the same track ripped at 192kbps or higher. :shrug:

      MP3 is a "lossy" CODEC. Fidelity is lost when creating the MP3 file. There are other "lossless" CODECs that will not lose quality. FLAC, raw wave, AAC, etc. are lossless CODECs. If you're a true audiophile, you may want to consider getting your music in one of these formats.

      Also, the higher the bit rate, the larger the music file. That's something else to consider.
      Have you looked in the FAQ yet?
      How about the Wiki?

      Under normal circumstances, a signature would go here.


      • #4
        Thats what I was wanting to know....what kbps is considered good quality. I hate the sound of MP3's and prefer WMA. As far as size that doesnt bother me, I have 2 terabytes sitting here doing nothing I can toss in. Im running Centrafuse wich uses Windows Media Player 11 to play audio, I think so, If that helps in answering my question there it is.
        Thanks for the input, before I go look now I am going to ask a question that may be real stupid but might as well ask while I'm here. If I click on properties of my music folder will it show me its bit rate? That way I can easily go thru and delete the bad ones.
        Thanks guys and for all you true audiophile and car audio guys by next summer I will have something that will make you all smile from ear to ear. Heres a taste........True real time 8 to 800 band eq control with 8 channels of crossover with high pass, band pass and low pass selectable at any frequency you desire.
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        • #5
          If you go to the folder and select View --> Details, then View --> Choose Details then check the Bitrate option, you can see the Bitrate inside the folder. That should make it easier to spot the low-quality files.
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          • #6
            right on.....thanks. Now I have something to do today.
            Google my name if you need to know who I am.


            • #7
              WMP is not the best player for audio quality.
              Have you looked in the FAQ yet?
              How about the Wiki?

              Under normal circumstances, a signature would go here.


              • #8
                have you looked into FOOBAR2000 with the SRC upsampler plugin? it does an outstanding job for me. I understand you use Centrafuse, and Darque is absolutely right when he says WMP is definately not for audio quality. also, have you tried ASIO4ALL? that will bypass your K-Mixer in your windows environment which should also clean up your sound a tad.

                To give you an idea of how my software is set up for audio, I use iTunes with multi-plugin so i can use the FOOBAR2000 sound engine, with the SRC plugin, I also use my soundcard for ASIO output...most of my downloaded rips are between 96 and 192kbps, and it varies quite a bit in quality, so i know the pain you are going through.

                good luck
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                • #9
                  Also mp3's do compress, but they can be non-lossy if the right parameters are chosen. However once you get there, the filesize is about the same as a nonlossy codec.

                  Every mp3 data point is stored in 2 bytes which is 16bits.
                  Human ear can hear generally between 20Hz and 20Khz. Nyquist sampling theorum says that if you sample at twice the maximum frequency, you can reproduce the original signal EXACTLY without any aliasing. So that is why the industry standard is a sampling frequency of 44.1Khz, which is 44100Hz.
                  Generally you have 2 channels for stereo sound (a left and a right), so only half your bandwidth is going to any one given channel.

                  16 bit/sample 44100 samples/second 2 channels = 1,411,200 bits/second

                  1024bits in a kilobit (notice these are in bits, NOT bytes as "kbps" is "kilobits per second" NOT "kilobytes per second")

                  1411200/1024 = 1,378.125 kbps for lossless audio

                  So you would need an mp3 with a bitrate of 1379kbps to reproduce exactly what is heard in real life. I think this is actually what CDs store, not 192kbps.

                  However, it is utterly useless to store that much info IMHO. Even a flute only gets to 2.5KHz maximum, and its little annoying baby the piccolo only gets to 5Khz. Some keyboards can make it up to the higher end of 8Khz or so, but I dont think anyone has every recorded something that annoyingly high. Human singing is generally between 70Hz for the deep deep voices and 1.1Khz for the high voice.

                  So unless you like squeaky music which you probably dont looking at all the subs in your cars , you probably dont listen to anything higher than a maximum frequency piccolo and even then that's probably a little high... So lets say 5Khz maximum music frequency, so 10Khz sampling frequency. Using the same math as above, you get: 320,000bps, which is 312.5kbps.

                  And that would make sense why mp3s generally only go up to 320kbps. Keep in mind that using anything above 312.5kbps encoding, you can 100% accurately reproduce (NOT just sort of kind of maybe with some errors blah blah blah, but truly accurately reproduce frequency for frequency) anything between 0Hz and 5Khz.

                  So then that asks what you can get out of other bitrates:

                  128kbps ~= 2Khz max
                  96kbps ~= 1.5Khz max
                  64kbps ~= 1Khz max

                  So for most instances, a 320kbps encoded mp3 will perfectly reproduce sound. Some exceptions being a birdcall CD that gets up to 14Khz, but then your speakers werent designed for that anyways, so it wouldnt work

                  And 2 minutes at 320kbps is 4.7 megabytes of storage space. Same 2 minutes with lossy encoding at 1378kbps is 20.2 megabytes. And there are no benefits. And for kicks, it would be under a meg at 64kbps if you needed to carry it around on a floppy

                  This is coming not from being an audiophile as I am not, but an electrical engineer that studied communication (as in modulation and demodulation of signals many many many different way) including how to compress them and get out what you put in. That sort of thing.
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                  • #10
                    nice explanation 2k1-- and from a audio standpoint it mostly lines up, though i have ripped at 64kbps wma/mp3 format in my earlier days (look how small the files are!), and there is a noticable loss of quality on the entire spectrum(the bass is 'muddy'-- kick drums have more of a thwonk sound instead of a thump)-- i have ruined 3 linkin park albums that still kindly remind me of quality to this day(some day, i will re-rip them).

                    most of my audio is ripped at around 192kbps, but there is still some noticable quality loss on certain tracks, so some day i wil be ripping my entire collection with a less lossy sound.

                    the hardest thing to capture on certain track is the ambience-- this is part of what gives music its depth--and is the easiest to loose in ripping.
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                    "The Project That Never Ended, until it did"

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                    • #11
                      I come at this issue from a different set of priorities. I want my ripped music to be the highest quality possible because this is a library of music that I will maintain my entire life and pass on to my kids. I rip everything in FLAC on a well equipped home PC with tons of storage. I use a program called Media Monkey to rip and organize the collection. Media Monkey also has the ability to automate the process of converting the music I select to lossy formats for loading on MP3 players, CD's, and DVD's. The user interface is not the best but the program features are very good.

                      Bottom line for me is the initial rip of a CD has to be highest quality because once converted to a lossy format, there is no going back to the original.


                      • #12
                        I rip stuff to FLAC for archival (and eventual media server) purposes, but also to the highest quality VBR (variable bit rate) MP3 setting using LAME. VBR only uses as high a bitrate as is necessary at any point in the track. It usually results in a substantial saving in file size compared to a fixed-bitrate MP3, with no loss in quality. I've only found one hardware MP3 player that balked at VBR, the Genica MP3 CD discman.

                        You should also insist on using a ripper that verifies quality using the AccurateRip database. EAC and DBPoweramp are a couple. CUETools can verify wav or flac files against the AR database, and tries harder than EAC to find the correct offset.

                        With files acquired from "other" sources, quality rips should be accompanied by a log file that confirms there were no errors in ripping. Choose albums encoded in FLAC, APE, or other lossless formats if possible. (If you get an album that's compressed as a single track with cuesheet, and want to split it into tracks, CUETools can split it.)

                        Just one thing, though. Bitrate and bandwidth are not directly related. A 20 kHz sine wave could be encoded perfectly adequately with a very low bitrate MP3. More complex high frequency waveforms like cymbals require much higher bitrates to encode transparently.


                        • #13
                          Here's my two cents...

                          MP3, no matter what bitrate is chosen, is lossy...plain and simple. There will be data removed and/or compressed when using it.
                          Upsampling, despite whatever quality your starting off with, will actually improve the sound reproduction...
                          Imagine an analogue sound wave...nice and smooth and rounded...a 'wave'. When digitally recoded (to CD), this wave is sampled along the waveform at certain 'points'...sample rate. 44Khz sampling takes a sample of the audio 44,000 times a second.
                          Now being digital, there can only be 1's ans 0's...on or off. The sound is reproduced not as a smooth wave form, like the original analogue audio, but something that more resembles a step ladder...the lower the sample rate, the bigger these steps are.

                          Now if you upsample this exact same data to a higher sample rate, you add more 'steps' to the wave form. The more 'steps' you use, the closer you get to a natural analogue sound wave..and of course, this will sound more like an actual analogue sound wave. (this of course isn't taking in to account the amount of data that is stored for each sample...either originally, or after upsampling).

                          The first few pages of this excellent pdf illustrate how this looks like...

                          As for all my music? Being a Windows based PC, I convert all my audio to WMA Pro lossless...whether it be losy based MP3's, or lossless FLAC (which constitutes 90% of my library), WMA audio files work very well in Windows and properly store all the id3 information.
                          I also use Centrafuse, so I'm also limited with the playback engine as well...I would prefer to use Foobar with SRC upsampling and ASIO with a VST crossover network...but CF3 still won't allow the use of different playback 'devices'.


                          • #14
                            it may be lossy in that it doesnt convert every time value to discrete time which would be impossible, but it can encode and decode with an exact reproduction of the source. Look up the Nyquist theorum. It doesnt make any sense to store every data point when fewer makes the same output.
                            Fusion Brain Version 6 Released!
                            1.9in x 2.9in -- 47mm x 73mm
                            30 Digital Outputs -- Directly drive a relay
                            15 Analogue Inputs -- Read sensors like temperature, light, distance, acceleration, and more
                            Buy now in the Store


                            • #15
                              Hi, I'd like to use foobar to centrafuse 3. Is it posible now? Is there an application or plug-in that allows this function?