Can I Change the Bit Rate of an Mp3?

I have seen this question come up a lot in the search terms that people use to get to Rooster’s Rail. So much so that I decided to answer the question here.

The short answer is no.

So I guess that deserves an explanation. Mp3 is an audio format that uses complicated compression formula that results in a much smaller file but retaining audio quality based upon what bit rate and sample rate was used to encode the file. Simply put:

Smaller file size = less quality

That is lower bit rate and sample rate.

Bigger file size = better quality

Higher bit rate and sample rate

Then there are mixes of the sample rate and bit rate that produces results that are in-between. There is also a file format called VBR or variable bit rate. I am not going to go into that here but that is basically where the bit rate varies according to the complexity of the music or audio file and results in a file size that can be smaller but retains a higher quality than an otherwise static bit-rate.

This bit rate and sample rate are dictated by the person that encodes the original file. The original file is often a .wav file wich is very large but of high quality. For example The Global Geek Podcast as a .wav file and goes for about one hour is over 500MB. Once encoded to an .mp3 file is between 20 and 25MB. I dictate the bit-rate and sample rate when I encode the file.

So lets say that you get that file which is encoded at 64bit and a sample rate of 44100khz. You want to lower the bit rate to make the file smaller. Sure you could using a program such as RazorLame, decode the file and then re-encode it at a lower bit rate such as 44bps (bit-rate). But the resultant quality would be lower than if I had used the original .wav file and encoded it at 44bps. This is because the best quality that you have is 64bps, that is as good as it gets. The quality can get no better than that. So in conjunction with the fact that .mp3 is a lossy file format and will get worse in quality every time it is opened and closed or encoded you end up with pretty much crap.

Same can be said if you wanted to go up in quality and therefore bit-rate. If you had a file that was encoded at 64bps there is no way on earth to make it better quality than what it is. If anything going up in bit-rate will make it worse because it will very successfully highlight the imperfections that are a result of encoding something as an .mp3 file. When a file is compressed in this way it decreases in quality regardless of the bit-rate. It all has to do with the fact that to make it a smaller file you have to ditch some of the data that the original file contains. That said it would take a very good ear to detect the imperfections in a high bit-rate encoded audio file, but it is there.

Moral of the story is that if you have an .mp3 file leave it as it is. Unless you can get the original source file there is no way to increase or decrease the quality and maintain any sort of standard about the quality of it. I hope this clears up a bit of confusion that there appears to be out there.
I tried to think of a good analogy to use for this post and could not, but this is the lame one that I did come up with:

Trying to change the bit rate of an .mp3 file is like baking a cake and then deciding that you want to know how to make it so it has less sugar in it than what you originally put in it and still have the cake!

The Hardest Edit Yet

You guys are going to think I rabbit on about the podcast a lot. Well simple truth is that I do. Reason? Well I (we) have a lot invested in it. It is a labour of love admittedly, but when you spend so much time on it; it makes it matter more. This week that is especially true.

This week we did our first interview for the show. It was with Dick Hardt of Sxip Identity, the CEO. Skype was a complete bastard, for what reasons we do not know. In addition to that the audio was less than perfect. As well as that I was learning my way around some new software. All up this meant for me a huge job. It took hours and hours of editing, and re-editing. I had to correct the audio levels and make sense of garbled Skype noise to extract the content. That is an added step that I don’t normally have to worry about.

I edited the interview once and on a listen I thought I could do better. I originally edited the raw data in Audacity then on the second I tried out Sound Forge by Sony. The wave patterns are easier to read in Sound Forge I think. That meant the second edit did not take as long. But a lot of the crackles and peaks were taken out. The software is very, very powerful and that meant that I had to learn a lot of new techniques for doing things. The help files are great and that helped. There are a few things that I could not help or eliminate, like the alternating volumes that can be heard. That was a result of the fact that no compressor on earth could have made up for the level differences between us and Dick. But overall the result is great compared to what I had to work with.

Then came the task of throwing it all together. Sebastian and I recorded our bit last night. That went so well it was smooth as silk. We knew our stuff and Seb was a great asset (as always). At one point my web page would not load. It was my story and once Seb knew I was having trouble he just stepped in and took over, magic. It just worked. Then after we finished I started to edit our bit and exported it as a .wav file. I put all the components together in Acid Music Studio, again Sony. It kicks butt and a serious time saver.

Audacity does multi-track badly. So this was a nice change. Again a powerful program and I was pulling my hair out with some things. Like if you change a track from a one-shot as opposed to a loop it changes the way the audio sounds. If it is supposed to be a one-shot, it sounds like crap as a loop. What I did not know was that it automatically loads some sounds as loops. So I spent about an hour trying to find out what the go was with Sebastian sounding like a Darleck! The other thing that this label of loop or one-shot does is changes the tempo of a track. So I was adding audio and it was either too fast or too slow! Again, pulling my hair out but got the two problems sorted when I realised the difference between a loop and a one-shot. Over all though I would say that software is very intuitive and easy to use. Despite my issues, which were minor. Organising and arranging your work is a snap and the ability to zoom and scale is priceless. I can now fit over ten tracks on the one view, much easier.

In addition to this the two programs work seamlessly with each other. Need to edit a track while working in Acid? No problem just open it with a right click on the file name, edit it and close and keep working on the project in Acid, awesome! The only thing that it won’t do is encode the .mp3, for that I am now using RazorLame.

RazorLame worked wonderfully. The file was encoded at 64kbps with a 44100khz sample rate. Another first for the podcast. I figure that it is a toss up between file size and quality and marketing. I think a smaller file size might mean more listeners. My only criticism of the file was that some kooky shit happened in that there is a bit of an echo in the first half of the podcast but then the rest is fine. Not sure what that was about but I think it might have something to do with modulation. So I have decided that for the next podcast record we will under level the audio, this will give the final product some headroom and take out some of the fluctuating echo. I say this because the audio of the interview which was under-modulated is fine.

And that’s a wrap. It was indeed the trickiest podcast to edit and toped with that was the new software and methods. I am proud of what has been produced. This is why we have a vested interest and maybe you can see why we are passionate about what we are doing. It represents a huge investment of time and energy. The reward is for people to listen to it.

Head on over to The Global Geek Podcast to check out the show!

What Bit-Rate Do You Use?

Since my original post about bit-rates and podcasting, I have thought a lot about what bit-rate I should encode the Global Geek Podcast. So after a lot of thinking we are going to give 64bps at a sample rate of 44100khz. This should give a fairly good balance between file size and quality.

In my original post I did a very, very small survey on the podcasts that were on my PC at the time and I thought why not do a full on survey and get a better picture of what everybody uses. So this post uses a service that is new to me but maybe not to you called, ZOHO Polls.

This is a great site where you can create a great looking poll, people can come and vote and they can leave comments. We are not allowed to post the poll on WordPress, but if anyone wants to post it on their’s visit the poll and click the add to Blog link on the top right and copy and paste the HTML. A lot of these things use Java Script and we can’t use that because there are security concerns and it is an easily exploited and used to inject a modified script that might be nasty.

So sound off podcasters, I want your vote! If the format you use for encoding is not here then please choose the closest and leave a comment. I can modify the poll so if I a getting a common theme for certain formats I will add it in.

Click Here to Vote!

You Can Also Subscribe to the RSS Feed for the Poll Here

Podcast Bitrate Problem Solved

As people that regularly read my blog I mentioned that I was having some major issues in regards to the encoding of the .mp3 file for the Global Geek Podcast. I was unable to encode the file at a bit-rate of 64 and then a sample rate of 44100khz. Audacity refused to allow this combination even though the “project” file was in a sample rate of 44100khz. Rather Audacity encoded the .mp3 at 64bps but then adjusted the sample rate to 24khz. This sample rate as you would be aware is not compatible with web based flash players. So not very useful.

It would appear that the problem that I was having was not an isolated one, other podcasters have come across this problem as well. It would appear that the problem lies not with Audacity but with LAME. LAME is the piece of software that actually does the encoding not Audacity. The limitation lies in that software. I have however sourced a solution with the help of my mate Adam – code monkey and general good guy.

The solution was not to ditch LAME as such. Rather we got hold of some software called RazorLame. RazorLame adds a powerful GUI (Graphical User Interface) to the LAME engine. I am fairly sure that it also includes some other software that meshes with LAME and the result is a top piece of software. It is open source as well which is great. As usual the interface is not that pretty but very functional and who cares about what it looks like as long as it does the job and this does more than that.

RazorLame will not only encode file but can decode files and then re-encode. Not recommended though, as I have said before .mp3 files are a lossy format and the quality deteriorates on repeated writing. But this remains a handy feature, it might get you out of a tight spot if you have lost the original file and you need to encode it again for some reason.

The big feature for me was the fact that you can mix and match bit rates and sample rates however you wish to. Makes for some interesting possibilities. But the feature that I was wanting to take advantage of was that I can now encode a file at a bit-rate of 64kbps, and a sample rate of 44100khz (or 44khz – for short). This is great because this means that we can keep the quality of the show but make the file size a bit smaller. As an estimate this means that our show will average 20MB to 22MB for 45 minutes to 55 minutes in length.

To take advantage of RazorLame you have to export the file from your audio editing software of choice as a .wav, it is this file that you select in RazorLame to encode to an .mp3. Remember to make sure you have enough disk space for this file as an hour show the file will be over 500MB. You can ten select the bit rate and the sample rate and other features that you may want to utilize. Then hit encode, that simple.

This is just another step in your editing process and one that should not be that hard to do. With the big payoff, a small price to pay.

I hope that this helps out all those podcasters out there who have had the same problem. For some reason the answer was hard to find. I dropped the problem here and in The Global Geek Podcast Blog and no-one responded with an answer. When I Googled the problem, I got my own blog entry stating the problem! So I decided to give the answer here as well!

Global Geek Podcast is Posted

Very pleased with this show. I just finished editing and writing up the show notes last night. It really is a great show and I enjoyed editing it and putting it up there. Every show is not without problems though!

So the part of the show that is usually hard, the editing, was a breeze. The hard part was the encoding of the file as an MP3! I wanted to drop the bit rate down to 64kbps, which I did. It was great, I had a show that was about 22MB, went for 50 minutes and still sounded great! But for some reason Audacity decided to change the sample rate to 24khz. The “project” was a 44khz project. So it should not have changed on the encode. So I ended up chatting to a few people and I got some advice and I tried it.

I exported the project as a wav file. So then the whole audio was mixed down in one continuous track at 44khz. This was done because it was speculated that perhaps some of the tracks that were added were influencing the output on the encode (for some reason). Anyway, I have this giganormous file and I open it up in Audacity and check all the bit and sample rates and I have everything set to encode the MP3 at 64kbps and the sample rate of 44khz. I encode the MP3 and the same bloody thing happens. On the encode the sample rate gets changed from 44 to 24khz. This can be a real problem for some web based flash players, they don’t like it and it speeds up the audio making it sound like chipmunks. Think of it like changing the RPM on the old vinyl.

So I do battle with Audacity for over two hours and get nowhere. I read forums and web pages to try to find out what was going on. Most web pages were an explanation of the difference between bit rates and sample rates, which is a difficult concept. But that was not my problem! So I gave up and decided that I was thinking about getting some commercial software anyway.

So if anyone out there uses Audacity and encodes their podcast at 64kbps could they please let me know what I am doing wrong? Thanks in advance.

But that did not ruin the show, the show was great and we had heaps of fun with it. If you have not listened to the show before then head on over to the homepage at The Podcast Network and check it out. You can even listen to the show off the web-page on the embeded player. I would prefer it if you were to subscribe though.

We have some great guests coming up as well, so stay tuned for that; it should be good. We have left it a bit before we got guests and interviews going for the show. We wanted to get comfortable with the show and getting it right as far as the audio and the editing.

So go get some geek over at The Global Geek Podcast. Remember to send us some feedback and let us know how we are doing with the show, but be nice. We do not mind honest but just saying it is shit because you think it is; is not nice.

This is supposed to be fun after all, thanks to our friends and supporters your positive comments have made the world of difference.