Home |
Search |
Today's Posts |
#1
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of
44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. Which choice would the average person prefer? Assume normal hearing. Thanks, Norm Strong |
#2
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
wrote ...
Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. Which choice would the average person prefer? Assume normal hearing. Do you think there is a single answer for all types of content (music, speech, etc.) and for all situations? If there were, why would we need separate control over all those parameters (and even others)? |
#3
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
writes:
Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. But it's not like that, actually. You appear to be applying linear PCM thinking here to a concept that involves lossy compression. Your question isn't terribly well constained so it's difficult to answer. One thing worth noting though, if you reduce the front end sample rate to 22k, Nyquist demands that you have to say goodbye to anthing near 11kHz or above straight away by low pass filtering the source. Listeners will have a big problem with that if we're talking about a general musical source, for instance. Also, dropping samples isn't a good way to do it. If you just casually throw away half the samples yet your analog input wasn't low pass filtered adequate to ensure the Nyquist criteria was met, you are nearly guaranteed to be introducing new frequencies into your sampled material through aliasing. Listeners hate that too. There are lots of ways to get a file size in half. Choice of codec, parameters provided to the codec, variable bit rate techniques, encoding mono vs stereo... No one size fits all surely. What problem are you trying to solve? Best Regards, -- /"\ ASCII Ribbon Campaign | Todd H \ / | http://www.toddh.net/ X Promoting good netiquette | http://triplethreatband.com/ / \ http://www.toddh.net/netiquette/ | http://myspace.com/mytriplethreatband |
#4
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
writes:
Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. Which choice would the average person prefer? Assume normal hearing. Thanks, Norm Strong Stereo - Mono -- Martin Schöön "Problems worthy of attack prove their worth by hitting back" Piet Hein |
#5
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
""Schöön Martin"" wrote ...
writes: Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. Which choice would the average person prefer? Assume normal hearing. Thanks, Norm Strong Stereo - Mono Brilliant! :-) I've done it myself. Especially for speech. |
#6
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
Richard Crowley wrote:
""Schöön Martin"" wrote ... writes: Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. Which choice would the average person prefer? Assume normal hearing. Thanks, Norm Strong Stereo - Mono Brilliant! :-) I've done it myself. Especially for speech. In practice, mp3 stereo files are mainly mono (sum of L and R) and a small amount of difference (L -R), so switching to mono will typically only give a 25% reduction in file size. Ian |
#7
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
"Todd H." wrote in message ... writes: Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. But it's not like that, actually. You appear to be applying linear PCM thinking here to a concept that involves lossy compression. Your question isn't terribly well constained so it's difficult to answer. One thing worth noting though, if you reduce the front end sample rate to 22k, Nyquist demands that you have to say goodbye to anthing near 11kHz or above straight away by low pass filtering the source. Listeners will have a big problem with that if we're talking about a general musical source, for instance. Also, dropping samples isn't a good way to do it. If you just casually throw away half the samples yet your analog input wasn't low pass filtered adequate to ensure the Nyquist criteria was met, you are nearly guaranteed to be introducing new frequencies into your sampled material through aliasing. Listeners hate that too. There are lots of ways to get a file size in half. Choice of codec, parameters provided to the codec, variable bit rate techniques, encoding mono vs stereo... No one size fits all surely. What problem are you trying to solve? The file already exists. 128kb/s, 44.1k. The question is how to cut this existing file in half with the least effect on sound quality. Going from stereo to mono would constitute a big sacrifice. Cutting the upper frequency response to 10kHz would be much less bothersome. But is there a better scheme? Another question: If I convert the mp3 above to a wav file, and then convert it right back to the same mp3 I started with, will the file undergo any deterioration? Norm |
#8
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
wrote in message
The file already exists. 128kb/s, 44.1k. The question is how to cut this existing file in half with the least effect on sound quality. Going from stereo to mono would constitute a big sacrifice. IME, not really. Cutting the upper frequency response to 10kHz would be much less bothersome. Most MP3 encoders implement a soft high frequency cutoff as the bitrate goes down. Dropping the sample rate to 22 KHz is obviously more like a brick wall. But is there a better scheme? Don't use MP3. Other formats do better at lower bitrates. Another question: If I convert the mp3 above to a wav file, and then convert it right back to the same mp3 I started with, will the file undergo any deterioration? Yes, but probably not as bad as the drops to 22 KHz sampling or mono. |
#9
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
In , on 02/06/07
at 10:26 AM, said: [ ... ] Another question: If I convert the mp3 above to a wav file, and then convert it right back to the same mp3 I started with, will the file undergo any deterioration? Norm Try it! It's not expensive in terms of time or $$, and it's a much better lesson than reading posts from a bunch of grumpy know-it-alls. Every time a WAVE file goes through the MP3 CODEC, things will be thrown away and the file will get smaller. The first trip through for the original file will probably yield the best compression on a percentage of the original file size basis. Your tolerance to the audio degradation will tell you when to stop. In general I think that you will find that the best result will be from establishing the compression ratio at the outset rather than fuss with a chain of MP3-WAV-MP3 ..., but this is something that you should play with. A a general rule, if the original compression scheme was any good (and appropriate for the type of data involved), compressing a compressed file does not result in much, if any additional space saving. ----------------------------------------------------------- spam: wordgame:123(abc):14 9 20 5 2 9 18 4 at 22 15 9 3 5 14 5 20 dot 3 15 13 (Barry Mann) [sorry about the puzzle, spammers are ruining my mailbox] ----------------------------------------------------------- |
#10
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
Ian Bell writes:
Richard Crowley wrote: ""Schöön Martin"" wrote ... Stereo - Mono Brilliant! :-) I've done it myself. Especially for speech. In practice, mp3 stereo files are mainly mono (sum of L and R) and a small amount of difference (L -R), so switching to mono will typically only give a 25% reduction in file size. Not too bad in other words. -- Martin Schöön "Problems worthy of attack prove their worth by hitting back" Piet Hein |
#11
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
wrote in message . .. Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. You have 2 choices: 64kb/s at the same sample rate, or throwing away half the samples, to 22.05k samples/second. Which choice would the average person prefer? Assume normal hearing. The L+R / L-R MP3 encoding automatically switches to mono if there is no stereo content. If you don't want the directional cues, you can save some space by switching to true mono. However, switching to mono or dropping the sample rate will produce obvious perceptual losses. Switching to a lower encoded bitrate will produce more subtle perceptual losses. By choosing mono, or lowering the sample rate, you choose a fixed -- and non-negotiable loss in quality. By lowering the encoded bitrate, you let the MP3 algorithm make the tradeoffs adaptively, in real-time. |
#12
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
Arny Krueger wrote:
The file already exists. 128kb/s, 44.1k. The question is how to cut this existing file in half with the least effect on sound quality. Going from stereo to mono would constitute a big sacrifice. IME, not really. Cutting the upper frequency response to 10kHz would be much less bothersome. Most MP3 encoders implement a soft high frequency cutoff as the bitrate goes down. Dropping the sample rate to 22 KHz is obviously more like a brick wall. The perceptual coder doesn't assign the same importance to the octave between 11 kHz and 22 kHz as it does to the range between 0 Hz and 11 kHz, anyway, so dropping the sample rate to 22 kHz wouldn't produce the desired halving of file size. Francois. |
#13
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
"(null)" wrote in message
news:1170880604.289039@smirk Arny Krueger wrote: The file already exists. 128kb/s, 44.1k. The question is how to cut this existing file in half with the least effect on sound quality. Going from stereo to mono would constitute a big sacrifice. IME, not really. Cutting the upper frequency response to 10kHz would be much less bothersome. Most MP3 encoders implement a soft high frequency cutoff as the bitrate goes down. Dropping the sample rate to 22 KHz is obviously more like a brick wall. The perceptual coder doesn't assign the same importance to the octave between 11 kHz and 22 kHz as it does to the range between 0 Hz and 11 kHz, anyway, so dropping the sample rate to 22 kHz wouldn't produce the desired halving of file size. Agreed. But, halving the sample rate should provide more size reduction than leaving the decision up to the coder. |
#14
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of 44.1k samples/second, and you want to cut the size of the file in half. Cut the song in half and keep the first part. :-) For example I see that on myspace.com-music people tend to listen to the first minute of the songs. If it's for a marketing thing maybe you don't need the second half of the audio track. F. |
#15
Posted to rec.audio.tech
|
|||
|
|||
Interesting technical question
|
Reply |
Thread Tools | |
Display Modes | |
|
|
Similar Threads | ||||
Thread | Forum | |||
I have a question | Audio Opinions | |||
Question regarding Phantom Power | Pro Audio | |||
Question regarding Phantom Power | Pro Audio | |||
newbie question - aardvark q10 + external mixer? | Pro Audio | |||
How To Write A Technical Paper By Eddie Runner | Car Audio |