Using iTunes. How to do I make mp3 files out of mp4 files - using cds?

Hi Marvin - thank you for confirming that!   I just now changed the setting and imported a cd that I had already imported earlier with mp4 files - I replaced the existing files, using the new setting.   Then I went to my iTunes music folder and checked the files, and they were MP3s.   Hooray!!!    It looks like from now on in they will all be mp3s.   That is fantastic…

@stepk wrote:

You are correct. The default encoder with itunes is aac. But, if you change it to mp3, it creates mp3 fles frm cd’s. The nice thing is ipods also play mp3’s. Many other players don’t recognize aac files. Maybe, Apple doesn’t give the authority.

 

Another free encoder that I love to use is bonkenc.

 

 http://www.bonkenc.org/

 

It rips cd’s really fast. I prefer 160 mp3’s. I don’t mind giving up little space for better quality. Remember to rip when you are connected to internet for tags.

 

Also, you need to choose where you want to save them.

 

Bonkenc also converts from aac to mp3.

The golden rule is avoiding to convert from one compressed method to another. Apparently, you lose some quality.

unless you have a lot of aac’s, re-rip the same cd’s.

 

Good Luck. 

 

 

Hi Sepk, that all sounds really good.   I have a whole stack of music in mp4 format, that I would obviously like as mp3s.

Now, a problem…   I am very ignorant, and I am going to have to be talked through this a bit.   

I went to the bonkenc site and found a link to an excellent tutorial about using it to convert audiobooks to mp3 files - but I have been able to find no tutorial about using it with music files.   The tutorial is better than nothing, but it leaves a lot of gaps re converting music files.

You say you prefer 160 mp3s.   When I go to iTunes Edit>Prefs>General…Import Settings… The setting I have there is given as "High Quality (160 kbps).   This sounds like what you are talking about…but I am not sure how this relates to using bonkenc.   

I would be really grateful for some more info.  

Thank you very much for the recommendation BTW…it sound like just what I need (if I can learn to use it!)

Whoops, sorry Stepk, I hadn’t finished…

You say “The golden rule is to avoid converting from one compressed method to another” I don’t understand what you mean by this…or what settings I would need to use in order to ensure I did this properly.

You also say “unless you have a lot of aac’s re-rip the same cds” - I’m sorry, but I don’t understand that either.   I do have a lot of mp4files on my computer…that is for sure.   It would be difficult and costly to re-do them again from the original cds (if that is what you meant.)  

I would be most grateful if you could explain more please.

@poppy wrote:

 

 

You also say “unless you have a lot of aac’s re-rip the same cds” - I’m sorry, but I don’t understand that either.   I do have a lot of mp4files on my computer…that is for sure.   It would be difficult and costly to re-do them again from the original cds (if that is what you meant.)  

 

I would be most grateful if you could explain more please.

 

 

 You shouldn’t convert MP4 files to MP3, since that will result in lower quality.  For best results MP3s should always be created from the original CD, or a lossless copy of the CD (such as flac, etc).  Depending on how picky you are, converting your mp4s may or may not sound ok to you.

FWIW, you can play MP4 files on a Clip+ if you run rockbox on it.  I’ve put a bunch of work into our AAC decoder and it works pretty well.   

Hi saratoga, thank you very much for that.

Hummmm, I checked out your website, and doesn’t it say that for sansa clip plus you have to be an advanced user to work it?   I am not an advanced user.   I am the pits…   I would need to be walked through even the most simple of programmes.   I am really ashamed to say that, but it’s the truth :frowning:

I am extremely sorry to hear that MP4 files should not be converted to MP3s.   I was so hoping to do that!   It sounded the perfect answer to my problems…    Maybe I will just experiment with a couple of albums, and see how I feel about the quality.  What a drag…

From the sound of it I may just need to settle for taking on future music as mp3 files.   I will probably have to settle for listening to my existing mp4s just on my computer…

@poppy wrote:

 

Hummmm, I checked out your website, and doesn’t it say that for sansa clip plus you have to be an advanced user to work it?   I am not an advanced user.   I am the pits…   I would need to be walked through even the most simple of programmes.   I am really ashamed to say that, but it’s the truth :frowning:

 

The port to the Clip+ has moved very quickly.  Soon we’ll have a simple installer that most people should be able to use.  Maybe check back in a couple weeks if you still want to play those MP4 files.  Right now we keep breaking things so its not a good idea for you to try rockbox until everything is working I think.

Hi saratoga…   Okay, will do!   Thank you…

Here’s what I mean Poopy;

The best sound on mp3 players is prvided by lossles ( huge files!) and flac (relatively big but not huge!) files. Obviously, very few people can notice the difference. To me, acceptable level of sound quality is what matters. You won’t notice a big difference between 160 and 256 kbps files. (Assuming you are using the player with a decent earphones.) Now, if you attach the digital player to your amp, and use your speakers, there is a diiference!. Only do you decide what is the purpose to use that mp3 player.

To my ears, 160 kbps is good enough. When I purchase dgital files from Amazon, they are 256 kbps. They sound excellent. When I convert these mp3 files into cda (compact disc audio?) and create a CD they are almost store purchased CD’s.

You are giving up certain sounds during compression process. Let’s say you rip a cd, and create 128 kbps aac’s. Now, you compress those again into 128 kbps mp3’s. It is not the same as those 128 kbps ripped from the original CD’s.

I won an ipod years ago. So, I naturally converted my CD’s into aac files. Then, I used itunes to convert them into mp3’s. (almost 32 GB!). I could tell the difference! New mp3 files were still good, but not as good as the aac files. I was afraid to break the CD tray on my laptop because I ripped so many CD’s.

You can use pretty much any music software to convert audio cd’s into compressed files. This is what I do with Bonkenc:

Launch Bonkenc. Click options/general preferences. under Encoder, choose Lame MP3 encoder. ( You will notice you can even use mp4/aac encoder for your ipod.) Click Configure encoder. under VBR mode, choose CBR (constant bit rate) under bit rate, decide what kbps you want. I chose 160. slightly better than 128. Remember that you are sacrificing more space on the player, the bigger the files are. click OK.

Under output directory, choose where you want to save them. Mine is my documents/my music/mp3 files.

Under filename pattern, choose what you prefer. I suggest artist/album/track/title

under general settings setup, click info tag. Make sure there are X’s in the small boxes next to Write ID3v2 tags and the last two. Click OK.

When you are connected to internet, pop a cd you want to convert. Wait until CDDB (compact disc data base) site shows the names of all the songs, band, album name etc. Very rarely are some info wrong! Users provide these info. Make sure they are correct.

Click Browse at the bottom of the bonkenc window, and create a separate folder. I prefer say Rolling Stones-Sticky Fingers.

This file name is for the computer. the digital player will recognise all the info.

Then all you have to do is drag and drop these files into music folder of Clip.

Good luck.

Sorry, I forgot to mention. When you are ready, click the right arrow (at top), to start the encoding process. Duh!

this baby is fast. thanks to its creators!

Stepk, I swear by encoding using VBR (variable bit rate), in which the encoder adjusts the bit rate for the needs of the audio at the time.  A great sound/size adjustment/solution.

Miikerman, I am not quite sure what is the best method. vbr or cbr. I see your point though.  maybe it makes sense to use vbr with certain genre.

A great debate. VBR (variable bit rate) or CBR (Constant)?

I did little search. the following has been copied and pasted from different people:

**Constant Bit Rate (CBR) - the same bit rate is used to encode the entire file.

Variable Bit Rate (VBR) - Mp3 files are made up from 100’s of small audio chunks, called frames. Whilst encoding a VBR file, the encoder decides which bit rate to use for each frame. The bit rate can drop down to lower value when it is permissible (if there is not much audio going on), and switches up to a higher value when required. VBR files are not all good news though, because the bit rate is constantly changing many players have difficulty displaying correct track lengths. A standard exists where the track length is encoded into the first frame after writing, though not all encoders do this and the ID3v2 tag can mess this up.**

CBR better…

All transcoding between lossy formats will lead to further loss of quality.

Unless you want to lower the overall bitrate significantly, you shouldn’t bother. Or if you have a portable device that cannot handle VBR.

**CBR, each frame is a set size
VBR, using physoacoustic models the encodered determines how many bits are needed to encode each frame. Most encoders let you set a range of bit rates, however the best encoder LAME has a set of built in standard (–alt-present (standard/extreme/[insane]) that it can use. Standard and extreme are both VBR with outputs that range around 220 for standard and 240 for extreme. Insane is 320CBR with high and low pass filters applied iirc. When a bit range limit is applied the encoder will only go down or up to that bit rate even if the frame needs less or more bits to be encoded properly. For this reason aps and apx uses a bit range of 32-320 (the “standard” mins and max for MP3 encoding).

The prolbem with VBR is that poor coding in windows prevents it from reading the bit rate correctly (winamp can read it correctly) and a lot of portable players cannot interpert the time correctly. They see the first frame (or an incorrect repersentation of the avg bit rate) as being something like 32kbps and then divide the total size (say 6MB) by that to get the (incorrect) time. While this shouldn’t/doesn’t affect the playing of files it is anoying. Overall VBR still will give you better sound quality for the storage space.**

In CBR (Constant Bitrate) encoding, the bitrate is kept constant across the entire file: the same number of bits is allocated to encode each second of audio, and internally, frames of audio data occur at regular, predictable intervals, so the overall file size for a given duration of audio is predictable. CBR is therefore the “opposite” of VBR.

That said, in some formats there may be some variability in the number of bits that contain actual audio information from frame to frame. This concept manifests in the bit reservoir of MP3s. In a CBR MP3, even though the frames are of a fixed size, the audio data is not necessarily distributed consistently between them; audio for one frame might use fewer bits than the frame has, so that frame ‘adds’ the spare bits to a ‘reservoir’ that can supplement the bits allocated to the next frame. Thus, the effective bitrate is allowed to vary somewhat in a CBR MP3, even though there is a fixed number of frames for the duration of audio. The bitrate of a single frame can be up to 320 kbps, but the frame that immediately follows that one would have to use fewer bits, whereas in VBR, there would be no such restriction. Consequently, the amount of variability across the entire MP3 is not as great as that afforded by VBR, but it is not insignificant; a CBR encoder that does not efficiently use the reservoir will likely produce a lower quality file than one that does.

Unlike in VBR, the perceived quality of decoded audio will tend vary across a CBR file. This is because CBR encoding is similar to the ABR form of VBR encoding in that it is normally based only on a target bitrate and analysis of the input audio; there’s usually no attempt to use the absolute lowest possible bitrate at which a particular output quality level would be maintained. Technically, CBR implementations always do incorporate a prediction of output quality, but it is based on fixed algorithms rather than trial-and-error testing of actual results as is done in VBR.

Who should use CBR

  • CBR is useful for people who are concerned about maintaining maximum compatibility, especially with certain streaming applications and some hardware-based decoders that don’t reliably support VBR.
  • CBR is also useful for people who desire the ability to obtain accurate estimates of the bitrate or approximate duration of a file’s decoded audio without scanning and partially decoding the entire file.

Advocates of VBR, especially on the hydrogenaudio forums, are often very vocally anti-CBR, and often say that no one should ever use CBR, when given the choice. Some reasonably argue that the point of using a compression algorithm, especially in a lossy codec like MP3, is to conserve as many bits as possible while maintaining a certain quality level, so CBR’s tendency to use more bits than is necessary in simple passages and to use too few for complex passages is wasteful and bound to produce worse results (in the complex passages, at least) than VBR. The fact that CBR implementations rarely take actual, rather than predicted, output quality into account is pointed to as another reason to avoid CBR.

However, these arguments need to be carefully qualified in order to be meaningful, and it would be incorrect to infer that there are inherent quality differences between CBR and VBR.

In general, however, for most types of input, assuming identical input, identical encoding methods, and sensible targets for VBR quality and bitrate bounds, VBR will almost always produce equal or better perceived-quality results than CBR for files of the same size or average bitrate , and this has been demonstrated in numerous double-blind listening tests. For example, using the same encoder, a 128 kbps CBR MP3 will almost never sound better than a VBR MP3 that averages 128 kbps, because in VBR, the simple parts of audio can be better compressed than in CBR, thereby allowing more bits to be available for the complex parts. On the other hand, since the simpler parts of the file sound better in the CBR version and the complex parts will sound be better in the VBR version, comparing even similar-bitrate files can be a very subjective experience.

CBR can exceed the quality of VBR if the comparison is not constrained to an average bitrate, or if the VBR encoding method does not take into account actual output quality. For example, a 256 kbps CBR MP3 containing moderately complex audio is likely to sound noticeably better, overall, than a similarly-encoded VBR one that averages 128 kbps, even though the VBR one may use up to 320 kbps in some frames. And even when VBR does measure output quality, there is a margin of error, especially when relying on perceptual psychoacoustic models, so the encoder (even the much-revered LAME) can accidentally overcompress some segments, depending on the characteristics of the audio, the quality and bitrate constraints imposed, and the capabilities of the particular encoder. At high bitrates, the quality difference between typical CBR and VBR files approaches zero, so, for some users, CBR is perfectly acceptable, especially if maximum conservation of space is not a concern.

At low average bitrates, the quality difference between CBR and VBR is more pronounced, given the same input, so VBR is often more desirable for applications that need a great deal of compression.

If input need not be the same, then VBR also makes it possible to keep the same approximate quality level as CBR but increase the frequency range of the input, which is often considered an increase in perceived quality even though there may be just as much quantization noise. For example, a ~96 kbps VBR file could use a 12.5 kHz lowpass filter on the input and have about the same percentage of noise as a 96 kbps CBR file with an 11.5 kHz filter. Depending on the listener’s sensitivity to noise in the additional upper 1 kHz, a higher overall quality level would likely be perceived due to the mere presence of those upper frequencies (assuming they contain audio that the listener wants to hear).

Constant bit rate (CBR) encoding persists the set data rate to your setting over the whole video clip. Use CBR only if your clip contains a similar motion level across the entire duration.  CBR is most commonly used for streaming video content using the Flash Media Server (rtmp)

Variable bit rate (VBR) encoding adjusts the data rate down and to the upper limit you set, based on the data required by the compressor. VBR takes longer to encode but produces the most favorable results.  VBR is most commonly used for http delivery if video content (http progressive)

VBR is superior. CBR is 20th century tech.:wink:

Marvin_Martian wrote:
VBR is superior. CBR is 20th century tech.:wink:

There’s a lot of 20th century things (tech included) that I actually prefer over the latest & greatest.

Sometimes Old School Rules! :wink:

Tapeworm wrote:


Marvin_Martian wrote:
VBR is superior. CBR is 20th century tech.:wink:


 

There’s a lot of 20th century things (tech included) that I actually prefer over the latest & greatest.

 

Sometimes Old School Rules! :wink:

You know how that makes you sound?

:dizzy_face::smileyvery-happy::stuck_out_tongue: 

Marvin_M;

Have you tried to rip the same song with vbr and cbr at the same bit rate? I’d like to know the difference with file size? can you hear the difference? 

I am planning to do this when I get a chance. Apparently, vbr gives better results with Lame Encoder’s improvements.

Stepk wrote:

Marvin_M;

 

Have you tried to rip the same song with vbr and cbr at the same bit rate? I’d like to know the difference with file size? can you hear the difference? 

 

I am planning to do this when I get a chance. Apparently, vbr gives better results with Lame Encoder’s improvements.

Can I hear the difference? No. As far as a size comparison, that all depends on the music. A single person singing and playing an acoustic guitar, that file would be a file that could save a lot of space, in comparison, a heavy metal tune with three guitarists wailing away, along with a singer and drummer and bassist, the difference would not be so great. My computer is acting a little bit temperamental today, or else I would do a couple conversions to give you an example.

Marvin_Martian wrote:

You know how that makes you sound?

:dizzy_face::smileyvery-happy::stuck_out_tongue: 

 

Wiser? More experineced? Worldly? Guru-like?

![](file:///C:/DOCUME%7E1/Paul/LOCALS%7E1/Temp/moz-screenshot.png)

Respect 'yer elders, you young whipper-snapper!

And get me another beer while 'yer at it. :stuck_out_tongue:

Marvin_Martian wrote:

You know how that makes you sound?

:dizzy_face: :smileyvery-happy: :stuck_out_tongue: 

 

Wiser? More experienced? Worldly? Even Guru-like?

Respect 'yer elders, you young whipper-snapper!

And get me another beer while 'yer at it. :stuck_out_tongue:

When I first started ripping my CDs, I tested different bitrates and methods.  I found that MP3 lame VBR produced the best sound for me in the least amount of space, in the format I then wanted (MP3).