Any audiophiles here? |
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cobb2
Forum Senior Member Joined: November 25 2007 Location: Australia Status: Offline Points: 415 |
Posted: July 04 2009 at 06:17 | ||
I think this line may sum up the audiophile problem with MP3. They know audio data is missing and may be pychosematic (bad spelling) to what they are hearing- it can't be as good can it? therefore it isn't
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 04 2009 at 06:01 | ||
For a typical 16bit/44.1khz WAV file the bitrate is 1411kbps, and that's what Windows will display to you. The important thing is that an MP3 file at for example 256kbps is not just a downsampled version of the original 1411kbps WAV file. The reduction in size is achieved primarily by looking at the frequency distribution of the signal and leaving out those parts that the human ear/brain can't perceive because they're masked out by other parts of the signal. Of course there are limits to this method of reducing size ... whether it's successful depends on many factors, most importantly the complexity of the signal and the resulting bitrate. If you encode a file at 128kbps, for most signals too many parts will have to be removed in order to accommodate the small file size. 128kbps is less than 10% of the original bitrate. If you encode at 256kbps (about 18%), you usually won't be able to hear a difference, or in other words: The mp3 encoder operates in a range that allows it to only remove parts of the signal that we can't hear, while still achieving the desired bitrate/file size.
I think this is the real problem: Some people simply refuse to believe that if you remove 82% of the original, the result can still sound exactly like the original (judged by human ears). Take into account though that even when you apply only lossless compression, most tracks can be reduced to about half their original size. Or let's say 60%, to be on the safe side. So adding the lossy compression to the equation actually only adds an additional ~ 40% of compression. You can also take into account that today tracks are usually encoded using VBR (variable bitrate). This means that for each frame the mp3 encoder can analyze it and decide - depending on the complexity - which bitrate to use. The encoder can also select bitrates for left/right channels separately. In a nutshell: There are many ways the mp3 encoder can use to optimize the compression rate. The bottom line for me: I have yet to hear a properly ripped ~256kbit MP3 that I could tell apart from the CD. I haven't done any real double blind tests, but I *never* found myself listening to such MP3 files and thinking "that sounds distorted/harsh/hollow, let's put in the CD". |
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cobb2
Forum Senior Member Joined: November 25 2007 Location: Australia Status: Offline Points: 415 |
Posted: July 04 2009 at 05:42 | ||
It's alright, I have figured it out. Because MP3 (read MPEG) was made especially for the web this is the streaming rate across a network, so on a wav file, windows is just reporting the stream rate.
addition: although it still makes sense that the lower the kbs, the less samples there must be within the streaming frames.
Edited by cobb2 - July 04 2009 at 05:48 |
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cobb2
Forum Senior Member Joined: November 25 2007 Location: Australia Status: Offline Points: 415 |
Posted: July 04 2009 at 05:08 | ||
You guys know far more about this than I do
The sampling rate I mentioned was bringing up the properties on a wav audio file in windows. The details always show a bitrate. This is what is misleading me?
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 04 2009 at 04:59 | ||
^ Yep - that's the page that mislead me
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 04 2009 at 04:52 | ||
You mean image 2-3 on this page: http://oreilly.com/catalog/mp3/chapter/ch02.html That image is actually not correct. It shows the effects of quantisation at different sample rates, which - as Dean also said - are not to be confused with bit rates of MP3 streams. The internet ... sometimes even O'Reilly can be wrong. EDIT: They're actually clarifying the difference later on ("Bitrates vs. samplerates") so they're not actually wrong, but merely inconsistent. Edited by Mr ProgFreak - July 04 2009 at 04:57 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 04 2009 at 04:46 | ||
CD sample rates are 44,100 snapshots per second, not 2000-3000. Sampling an audio signal at 3000 samples per second would result in an audio bandwidth of 1.5Khz - which is about half that of an analogue landline telephone system and no where near good enough for music. The lost data in lossy systems is far to complex to describe in simple terms, but essentially the information that is lost is data that the human ear cannot hear because it is masked by more dominant sounds in the music. Mp3 codecs do not fudge the missing information - once removed it stays removed.
My error was confusing bit-rate and data-rate -
bit-rate = speed of the digital data stream in bits
data-rate = speed of the digital data stream in words (number of bits per sample x 2 for a stereo signal)
sample-rate = the speed in which the analogue signal is sampled
in uncompressed data the data rate is the sampling rate, the bit rate is 32 times greater than that (44,100 x 16 x 2 = 1,411,200);
in lossy systems the data-rate is not the sample-rate so are not related since (as Mike pointed out) the compression is done after sampling at 44,100 times a second - the resulting bit-rate of 128,000 bits per second is the compression from 1,411,200 bits per second - the original sampling rate remains unchanged.
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cobb2
Forum Senior Member Joined: November 25 2007 Location: Australia Status: Offline Points: 415 |
Posted: July 04 2009 at 04:37 | ||
Yes, but the bitrate still controls how much information in these frames is retained. That's why further on in this chapter a simple sine wave is shown displaying how a 64kbs retains less information than a 128kbs sample. I guess mine is a very basic view of the process- I deal with this with my Information Processes and Technology students (final year of high school) and they don't have to know the process in great detail. I basically just tell them that noises we can't hear are removed and show how the sampling rate works- using the sine wave example.
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 04 2009 at 04:26 | ||
^ thanks, this definition is actually more concise and to the point than the ones I found at Wikipedia and HydrogenAudio.
However, in your explanation it sounded like the mp3s were simply files with less information than the original ... like when you watch a DVD on the computer and reduce the size of the video window, which would be an analogy to downsampling (or re-sampling with a smaller resolution). That's definitely not the case with MP3 ... the key is that with MP3 you can preserve most - if not all - of the important information of the original signal. You could never do that with a downsampled version ... upsampling it will not restore the quality. |
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cobb2
Forum Senior Member Joined: November 25 2007 Location: Australia Status: Offline Points: 415 |
Posted: July 04 2009 at 04:19 | ||
Okay- mine was an attempt at explaining in a more understable way (I only mentioned resample, not up or down sample)
Here's an extract from O'Reily's Definitive MP3 MP3 uses two compression techniques to achieve its size reduction ratios over uncompressed audio-one lossy and one lossless. First it throws away what humans can't hear anyway (or at least it makes acceptable compromises), and then it encodes the redundancies to achieve further compression. However, it's the first part of the process that does most of the grunt work, requires most of the complexity, and chiefly concerns us here. Perceptual codecs are highly complex beasts, and all of them work a little differently. However, the general principles of perceptual coding remain the same from one codec to the next. In brief, the MP3 encoding process can be subdivided into a handful of discrete tasks (not necessarily in this order):
I don't doubt that you and dean will grasp this, but most won't. |
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 04 2009 at 03:15 | ||
Sorry, but that's not really an accurate description of what mp3 does. THERE IS NO DOWNSAMPLING OR UPSAMPLING GOING ON. What the encoder actually does is it takes short intervals of anything from ~100 to ~1200 samples (called an mp3 "frame") from the input stream and then analyzes the frequency distribution. Have a look at this page for further technical details: http://wiki.hydrogenaudio.org/index.php?title=MP3#Encoding_and_decoding In a nutshell, for each of these frames the mp3 encoding algorithm comes up with a shorter block of data , based on the frequency distribution data gathered. This data can then be used by the mp3 decoding algorithm to reconstruct a series of samples which is a reasonable approximation of the original (in that it sounds the same to the human ear). Edited by Mr ProgFreak - July 04 2009 at 03:46 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 04 2009 at 03:01 | ||
Yes... for once Unfortunately I picked a website that didn't
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 04 2009 at 02:59 | ||
The Wikipedia page explains it better than I ever could ... |
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cobb2
Forum Senior Member Joined: November 25 2007 Location: Australia Status: Offline Points: 415 |
Posted: July 03 2009 at 20:50 | ||
Sampling rate is just simply taking a snapshot of the frequencies at a given interval. CD usually lies somewhere between 2000 - 3000 snapshots per second. MP3 is whatever you set it up to sample at. 320 would resample the original wav at 320 snapshots per second- lossing 90% of the frequency data. This is gone forever, hence the lossy format. The magic occurs in the mp3 codec algorithms which attempt to fudge the missing information. The lower the mp3 sample rate the more impossible it is to fudge this. Hence a 64kbs is perfect for web download speed but quite atrocious to listen to.
addition: if you then burn the 320kbs back to cd the algoritms in the burning software will resample it back to the higher wav sample rate, but now there are just straight lines between the original 320 rate and the new 3000 rate. Edited by cobb2 - July 03 2009 at 20:54 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 03 2009 at 17:41 | ||
Because I didn't know and wanted to know the answer.
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 03 2009 at 17:34 | ||
I never said burning a 16bit/44.1Khz WAV to CD-R degrades quality (that is simply a storage media transfer) - I assumed (incorrectly) that the mp3 files had been upsampled during the mp3 encoding to achieve the differing bitrates and would therefore have been downsampled to produce the WAv file.
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 03 2009 at 17:20 | ||
Your logic escapes me ... and I'm too tired to explain it again. I'll check back tomorrow ... maybe you'll manage to explain to me how burning a 16bit/44.1khz WAV file to CD-R degrades the audio quality.
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 03 2009 at 17:18 | ||
^ You're right I am - I've confused bit-rate with data-rate of course a CD bitrate is 1,411.2Kbps.
However, that does not remove or negate my reservations over that test.
/edit: don't believe all you read on the internet: http://www.mp3-converter.com/mp3codec/bitrates.htm Edited by Dean - July 03 2009 at 17:21 |
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Mr ProgFreak
Forum Senior Member Joined: November 08 2008 Location: Sweden Status: Offline Points: 5195 |
Posted: July 03 2009 at 17:12 | ||
You're really wrong here. The psycho acoustic algorithm (the mp3 compression) is applied to the 16bit/44.1khz source. The result of this algorithm is the 128kbps stream. If - as you said - the signal was downsampled first to some combination of word length / sample rate that fit into 128kbps, the result would severely degrade ... and why would you need to apply any mp3 compression to that stream, if it already was at 128kbps? And like I said above ... even if it all was like you said, that would make the files more distinguishable from the original, not less.
Sorry, but you're contradicting yourself here. EDIT: It seems like you think of the mp3 data stream like it contains a series of samples ... it doesn't. It contains information that the mp3 decoding algorithm can convert into a series of samples ... which match the word size / sample rate of the original signal. Edited by Mr ProgFreak - July 03 2009 at 17:14 |
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Dean
Special Collaborator Retired Admin and Amateur Layabout Joined: May 13 2007 Location: Europe Status: Offline Points: 37575 |
Posted: July 03 2009 at 17:05 | ||
^ okay - the 128kbps bitrate is the datarate of the mp3 file - this datarate is split across both left and right channels, resulting in 64kbps bitrates for each channel - or 64,000 samples per second - this sampled data stream is then processed by the compression algorhythm to extract the psychoacoustic information - this means that the 44.1khz sample rate of the original signal has been converted to 64Khz sample rate - that is upsampling. On playback on an mp3 player that same sample rate (64K) is used to playback the music - converting it to a 16-bit 44.1Khz WAV for playback on a CD player is downsampling it. Any mathematical process such as up and/or down sampling between non-coherrant sampling frequencies will introduce errors that were not evident in the signal before conversion. That is my reservation with this test.
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