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>H.I._Seventy_Five.mp3
Goddamnit Grey. Now there's not even a number there!
Nothing is consistent, not even the encoding settings. The last episode was 256kbit/s, and this one is back to 128kbit/s. You should use the lame mp3 encoder, with settings such as this: lame -V4 file.wav
This happens because MP3 encoders add silence to the start and end of files, for technical reasons to do with the length of encoder frames. If you're interested in the technical details, they're explained pretty well on the LAME homepage.
Of course, many tracks just have some silence at the start anyway. If the silence is longer than a fraction of a second, it's deliberate rather than an artifact of the encoder.
There are a number of open source licenses that allow you to redistribute a work within a commercial product, such as the Apache license 2.0, the BSD license, the LGPL, and the MIT license, provided the license requirements are met.
LAME is currently licensed under the LGPL but I don't know if that applied in 2005.
Sony violated the LGPL by not acknowledging the use of LAME in their distribution. LAME's LGPL license did not require Sony to release the source code of their rootkit.
H.264 (MPEG-4 AVC) is a video compression format, x264 is a piece of software that encodes video into this format. Just like we had XviD and DivX before which were both MPEG-4 ASP encoders. What you're asking is like asking "is it better to download a LAME (http://lame.sourceforge.net/) release or an MP3 release" (if you are more familiar with audio), which doesn't make much sense.
I want you to do a blind test with a properly encoded MPEG-1 AL3 ( ≥ 245 kbit/s compressed with a quality encoder like "lame") versus a .flac of the same song (just download your favourite song as a .flac and convert it yourself using the lame codec set with audacity or any other music-editing tool). Then get another person and let them play both files for you while you are blindfolded / not looking. For an easy evaluation do 10 listen samples. You partner writes down if you guessed correctly or not. I bet my left nut and my right one as well that you can't get at least 7 out of 10 right.
Unless you spend an exorbitant amount of money on headphones and a proper ODAC you're not going to hear any difference! It's hard to believe at first given the big jump in file-size, but you can rarely make out a clear winner. It mostly comes down to how the music is produced, where the artists equipment has its limits and what hardware / software he uses to compose.
Joel know his stuff pretty well and I'm sure he goes for as small of a quality loss as possible when it comes to audio-convertion.
TL;DR: .flac is a waste of disc-space when compared to a proper .mp3 file. Like most PC-Audio related stuff, .flac is snake-oil.
I manage to hear the difference between FLAC and mp3 LAME 320kbps. Its not easy and personally i woudnt be sad if i was stuck with mp3 LAME 320kbps till the rest of my life. Its a small small difference that people who like to focus on minor details can hear. Not everyone is the same and not everyone has an audiophile DAC and headphones as well.
There are ABX tests you can take with foobar add on that prove to you that you can hear the difference and your "many studies which you dont cate to google" is bullshit.
edit: Im ready for the downvotes since i see it is strongly disagreed with to have good ears, good headphones and a good DAC. If there were no difference i guess every single audio producer, engineer or a musician are dumbasses for not using simple mp3s in their production instead of lossles.
Not just the bitrates, encoders got much better as well. A 128 kbit MP3 from a new MP3 encoder will sound much better than the same 128 kbit MP3 created by a 15 year old encoder.
Lame for example has a big sample of test files to tune their encoders to create even better output: http://lame.sourceforge.net/quality.php
Depends on the encoder/decoder.
Because of several reasons, audio that goes though an MP3 encode/decode process will be longer and out of sync. The delay/padding can be compensated when the encoder correctly sets the MP3_ACCURATE_LENGTH, ENC_PADDING, and ENC_DELAY tags, and the decoder uses them. It looks like this in foobar2000.
I tested this with lame 3.99.5 command line, both as an encoder and decoder. The decoded WAV has the correct number of samples and sync. The ffmpeg encoder has correct sync but is a bit longer. It seems the ffmpeg encoder puts the correct delay but sets the padding to zero, thus the decoder does not remove the extra samples at the end.
Some DAWs like Reaper use these tags when importing or exporting MP3s.
It is a standard mp3, with a variable bit rate ("vbr"), produced by the "lame" encoding engine (http://lame.sourceforge.net/).
The '0' indicates it is the best quality lame vbr preset.
-V0 (~245 kbps), -V1 (~225 kbps), -V2 (~190 kbps) or -V3 (~175 kbps) are recommended. These settings will normally produce transparent encoding (transparent = most people can't distinguish the MP3 from the original in an ABX blind test). Audible differences between these presets exist, but are rare.
Suddenly I feel like an asshat for bringing any of this up. I was just commenting on the original comment that said in blind test, most couldn't tell between FLAC and 320. And I can relate to this, as I see a lot of people complaining about wanting a 320 rip. And most in the audio community no longer accept 320 as THE BEST for mp3, yet a lot of people don't know this. So when something leaks in V0, they only want 320, and then they even think the 320 they get later on sounds so much better. But if you compare the 320 and V0, there is no difference. So it's just interesting to me how much the placebo effect is in place when it comes to audio formats.
It's interesting that so many people are concerned with wanting to be able to discern a difference between CD audio and a lossy encode of it. Lossy audio compression formats and encoders are designed to achieve perceptual transparency, so it's no real surprise that hearing a difference is often quite difficult, especially under normal listening conditions and using medium to high bit rates.
There are a number of factors that may contribute to one being able to (or not being able to) discern a difference between CD audio and its lossy counterpart, such as:
- Lossy audio format, encoder, and settings used in the encoder
- Physical hearing ability of the listener
- Mental and physical state of the listener
- Familiarity of the listener with lossy audio compression artifacts
- Characteristics of the playback equipment used
- Acoustic properties of the listening area/room
If you're really interested in trying to hear a difference, becoming familiar with common lossy audio compression artifacts should be the first step. See the Quality and Listening Test Information page on the LAME website for samples that LAME has difficulty encoding. Also search Google for more samples that LAME struggles with (the term "killer sample" will be useful), read up on lossy compression artifacts in general, and note that different lossy audio formats generally produce quite different sounding artifacts.
I would upgrade your speakers for reasons wholly separate to the topic of being able to hear lossy compression artifacts, but once you do upgrade, don't be surprised if you still find it difficult to hear artifacts in medium to high bit rate lossy encodes. You may not even be able to hear artifacts at all the majority of the time – but I would say that this would be a testament to the lossy format and encoder used, not an indication of a lacking playback system.
Atleast one of the Open Source projects that was infringed on was the LAME mp3 encoder, which is covered by one of the versions of the GNU Lesser Public License.(Fox News article about the use of LAME and several other open source projects in the root kit)
LAME's license (in at least it's simplest requirements for non-infringement) would have required Sony to clearly announce that we using code derived from LAME
LAME's short and sweet license FAQ
So, ignoring the debate on the finer points of whether some public licenses would have allowed the inclusion of the covered code without being in violation of the license that occurs below; here is at least one concrete example of Sony violating a license agreement.
Amusing side note, since LAME does not technically have licenses to any of the relevant MP3 patents, distributing a compiled version of LAME may have also put Sony in violation of some of the MP3 patent agreements
Ninja edit:
I misread some things and consider my defense invalid. Stupid US patent laws.
> Today, LAME is considered the best MP3 encoder at mid-high bitrates and at VBR, mostly thanks to the dedicated work of its developers and the open source licensing model that allowed the project to tap into engineering resources from all around the world. Both quality and speed improvements are still happening, probably making LAME the only MP3 encoder still being actively developed.
iZotope would have licensing issues with LAME?
>http://lame.sourceforge.net/license.txt
Nope.
I haven't read the new manual, but I guarantee Ozone will specify which specific codec you are listening to. They're pretty thorough. I just wish SoundCloud would go to lossless so I can call it a day. Their shitty audio quality drives me bonkers.
I would recommend using EAC (exact audio copy) with LAME for MP3 encoding. You can set EAC to automatically encode using LAME after extraction, and set the variables such as bitrate.
I don't know /u/gindc but it's possible that the reported difference is real but not due to the bit-rate. Certain types of sound are hard to encode as MP3 (or AAC, Vorbis, etc.) and a surprising number of files are still around which were created by ancient codecs with noticeable artifacting.
https://en.m.wikipedia.org/wiki/Pre-echo
(The LAME project has used various test cases over the years to detect and handle this kind of problem: http://lame.sourceforge.net/quality.php)
It'd be interesting to see if the current version LAME produced better results on those same files.
LAME only encodes to mp3. and the first track was released as a single so it's higher quality. FFMPEG, well ffprobe confirms they are (other than the first one): Stream #0:0: Audio: mp3, 32000 Hz, stereo, s16p, 96 kb/s
Thing is, you're talking about low bit rate MP3 encoding, not digital audio in general. Encode at a better bitrate e.g. LAME VBR 0, and you don't get those ruined cymbals and other artifacts.
Some artists or suppliers avoid that compression through the use of FLAC files that use only lossless compression e.g. I bought Steven Wilson's The Raven That Refused To Sing in 24/96 FLAC format. That's technically way beyond what the human ear can appreciate in both frequency response and dynamic range. Technical specs alone are no guarantee of quality music, but put this album through a quality D/A and studio monitors, and you're welcome to your vinyl.
Bad idea. If you want lower bitrate mp3s, start from the original files, not from an mp3, because you'll get worse quality in the latter situation than in the former.
There might be transcoders available that directly convert mp3 to a lower bitrate without actually decoding it to audio first (meaning: each fame of an mp3 file is turned into a new frame, just with fewer bits), but I can't name one such program from the top of my head.
I know lame (command line encoder/decoder, used by a lot of mp3 handling programs) can directly convert mp3 to mp3, but I found no evidence it it doesn't convert to raw audio in an intermediate stage -- even though it doesn't create any such temporary files in the process.
I'm not sure if this is just a joke or not, but that's not how metadata works. If the an MP3 decoder starts decoding the frame at the very beginning of the file where metadata starts, it would simple fail or you would hear garbage. Metadata is also completely optional and can be removed from a file without any consequences, as long as the required header containing the information about things like bitrate and the like are still intact. Also, metadata can be stored at the end of the file as well.
Having said that, I don't have answer as to why MP3s have some silence. I can definitely say it has nothing to do with the metadata, though.
Edit: Just looked it up since I was curious after making this comment. It's much more technical than metadata and I haven't read it entirely so I can't give a simpler response, but here's the real answer, straight from the developers of one of the most popular MP3 encoders: http://lame.sourceforge.net/tech-FAQ.txt
I suspected that audacity would do the trick, and a google search appears to confirm my theory.
-Audacity 1.36 beta with ffmpeg and Lame installed should be able to open a .wmv file directly and convert the WMA to MP3.
If the quality only sounds like crap when exported to Mp3, then maybe the problem isn't Audacity it's self.
The MP3 encoding on Audacity is done with the 3rd party Lame mp3 encoding library. Maybe you need to upgrade, configure, or otherwise fix lame.
Edit: You used to have to install Lame separately from Audacity. I don't know if current Audacity installer versions include Lame.
When I was on what.cd, a lot of the total audiophile heads swore by this program.. Exact Audio Copy - you will also need to go get LAME and add the binaries to your system, so the ripper will have the libraries to encode with. The LAME sites have decent instructions on where the .dll files go.
Here is a pretty good general overview.
The Wikipedia page is also pretty darn good. Gives some context.
Also, if you want to really dig into it, the LAME encoder is open source, and used everywhere.
foobar2000 with the LAME encoder binary (lame.exe) is one option. Put lame.exe into the main foobar2000 directory and the Converter provides a standard interface for LAME's VBR (variable bit rate) settings. You can also create a custom encoder preset to use other options, such as CBR (constant bit rate). You'll want to choose settings or command line options that optimise the size to quality ratio for your needs.
If using command line options in foobar2000, below are a couple of examples of strings you might use:
- --noreplaygain -V 2 - %d
- --noreplaygain -b 320 - %d
- --noreplaygain --abr 287 -h - %d
WAV is for mastering and is completely lossless. MP3's lose some quality.
If you are on Windows and if you do a lot of conversions and want a polished professional product, I recommend dbpoweramp - it's definitely worth it if you plan on ripping a bunch of CDs.
If you don't mind getting dirty with the command line though, you can just download the Lame Encoder and use it. I'm 99% sure that is what dbpoweramp is actually using for the conversions, they just put a nice friendly GUI in front of it.
tl;dr: If you use some free or open source software to create MP3 files from your own CDs for personal use, this will be perfectly fine.
Again, this are two separate questions:
1. Can I make a digital MP3 copy of the songs I bought on CD? I'm not really familiar with the US laws, but this source seems to say that you are probably fine when you do this for personal use.
2. How about the license of the LAME encoder? You don't want to infringe any rights of third parties, so I will you give you some examples of what is ok and what would be not:
- download a pirated copy of some CD ripping program which normally costs money (e.g. WinAmp Pro) and uses the LAME codec: not ok. LAME itself is free (LGPL license) software, but that does not mean that all software using LAME code are.
- download the LAME source code, compile it and use it as an encoder: ok.
- use a free or open source program which uses LAME (see this list for examples): ok.
Yes. It has a headphone jack, so that's good. The way I do it is like this:
- Connect the headphone jack on the cassette deck to the Line In (not Mic) jack on a computer. I use a simple 3.5mm audio cable, with a 3.5mm-to-0.25in adapter for the headphone jack on the cassette deck.
- On the computer, set the input level for the Line In jack using your system's audio settings utility. This might take some experimenting, based on your computer and your cassette deck. Too high of an input level will cause distortion in the recording, while too low a level will result in a weak sound. For the combination of my Technics M228X deck and my computer, an input level of 40% on the computer works fine.
- Open an audio recording program that can read the Line In input. I use Audacity, which is free and makes it easy to record and see the waveform of the signal during recording. I then save the recording to WAV format and break it into individual tracks; the waveform shows the gaps between songs (flast signals with no amplitude).
- Convert the individual tracks from WAV to MP3. I always do that separately from the previous step, since cassettes don't have any song metadata in them like CDs do. There are many ways of doing this. I'm a Linux user, so I use LAME on the command line, entering all the metadata myself.
Alright so I'm trying to install LAME now and I'm running into a few problems. I am trying to download using this site and am getting a .tar.giz file, which I frankly have no idea what to do with. I've tried extracting with 7-zip but haven't had any luck. Sounds like I should be getting a .exe from that site.
The majority of the functionality of this program is coming from system commands/shell scripts which are not apart of Perl, let alone Strawberry Perl.
I'm on mobile right now so I can't help much, but I'm almost positive you'd need to install "LAME" in addition to Perl.
I can't really tell, but it might be a UNIX program only. And since you're using Strawberry Perl, I'm assuming you're running Windows.
That being said, the script seems fairly simple and that it could use alternative programs/scripts instead of LAME & or Perl.
I'll take a look when I'm on desktop later
You can download the latest version of LAME from http://lame.sourceforge.net/ , but you most likely encode your mp3s from a frontend program (iTunes, MusicBee, etc.), in which case you just have to wait until they update the version of LAME they use
Foobar2k is a good one, as you can save conversion presets (like output directory, bitrate) for quick conversion.
And as always, mp3 encoding is mostly done using LAME.
Download as MP4, will come out as MP4-DASH, convert with foobar2000 via lame MP3 encoder.
I'm sure there are encoders for other formats as well, but I know for fact that this works.
A) Some versions of Vegas install packages get confused with MP3 codecs (I'm guessing you might have come across your install through less-than-legal means?), and I've had success downloading codecs manually, for example the LAME encoder.
B) <diatribe> I grew up using Vegas ever since it was Sonic Foundry Movie Maker or whatever. We grew together, and got to know each other quite well But finally, last year, I had enough and we went through a turbulent breakup. I got tired of system crashes, shit like this codec issue, poor support and update frequency. I wanted to get more engrained with an industry standard, so I took the leap in to Premiere. After a year of getting used to the quirks and shortcuts, I have no regrets. Sony didn't care enough about the brand and they've lost my trust. So, I guess, Viva la Adobe. </diatribe>
From my reply to his post:
Here's a table of LAME (the most popular and well developed mp3 encoder - in fact pretty much the only one these days..) bitrates and associated LPF cutoffs:
http://wiki.hydrogenaud.io/index.php?title=LAME#Technical_information
Here's LAME talking about their psychoacoustic model (GPSYCHO):
http://lame.sourceforge.net/gpsycho.php
>Lowpass filtering based on the compression ratio. For high compression ratios, low pass filtering will improve the results. The exact amount of filtering needed depends on the music and personal preferences - the formula to decide how much lowpass filtering to use may need some tuning. At 256kbs, no filterings is done. At 128kbs, the lowpass filter is around 15.5khz.
I see this all of the time. In my most recent video, for example, Audacity added exactly 20ms to the end of the audio stream that I use for my voice. However, when I compare the raw file to the processed file side by side in Sony Vegas, everything is perfectly in sync. What I usually end up doing is simply trimming the extra bit that Audacity added.
A quick Google search indicates that this has been a known issue since June of 2000.
"Why is a decoded MP3 longer than the original .wav file?
Because LAME (and all other MDCT based encoders) add padding to the beginning and end of each song. For an explination of why, see the questions below.
LAME embeds the amount of padding in the ancillary data of the first frame of the MP3 file. (LAME INFO tag). The LAME decoder will use this information to remove the leading padding of an MP3 file.
Modifications to the decoder so that it will also remove the trailing padding have not yet been made."
Here's the source: http://lame.sourceforge.net/tech-FAQ.txt
Long story short, it's just a quirk of the program.
Yeah, the plan was to run lame within Lambda, to encode WAV to MP3. But I wasn't sure if Elastic Transcoder can do this much more easily. The catch is, it has to be able to detect when a user uploads to a bucket, transcode, save to another folder in the bucket, and then let my Node.js app it's successful. I'm not sure if it does that. ET seems to be for pre-determined batch jobs.
That means the MP3 is not encoded properly.
Every working MP3 in the game is encoded with this
But that file isn't.
This is why you can't copy-paste MP3 into the game's directory, change the filename, and expect it to work.
(open the MP3 with Notepad for example and you'll see Encoded with LAME v3.97 somewhere)
So that's what you meant by trickery, and in that case you are correct.
On a related note:
Proper MP3 encoders (like LAME) dynamically switches between simple stereo and joint stereo, on a frame-by-frame basis. Disallowing use of joint stereo, only cripples the encoder and results in worse quality.
Hate to break it you bro, but you are totally wrong on this. The reason the developers of LAME chose their name and acronym is not because LAME is not an MP3 encoder. LAME has always been an MP3 encoder.
Denying that LAME is an MP3 encoder was part of a legal strategy employed by the developers of LAME. You see, as an MP3 encoder, LAME is infringing on patents held by Fraunhofer and others.
In recent years, these challenges have abated somewhat and LAME now outright states on their website that "LAME is a high quality MPEG Audio Layer III (MP3) encoder".
Please note that H.264 and x264 aren't the same thing.
H.264 is the standard, while x264 is a piece of software targeting the standard, there are other encoders around which perform worse, or just utter crap
Same rules apply to audio, a good MP3 encoder (LAME) will outperform a bad AAC encoder.
My general goto settings with ffmpeg are:
ffmpeg.exe -i <input> -c:a copy -crf 18 -preset veryslow -tune film <output>.mkv
Here's a quick sample of something I recorded yesterday, though excuse the audio its out of sync. I think its about 6mbps and encoded at 7.8fps on my Phenom II X6.
Edit: with respect to your question about downscaling, do it in post, as you can use a high quality downscaler without impacting performance -vf scale=1280:720:flags=lanczos,setsar=1/1 on the ffmpeg command line should do it.
You dont actually describe the problem itself at all.
something about converting mp3 but no info other than that?
cant write lame? you need the lame dll in the folder to exist. find the lame.dll, put it in appropriate folder, its not that hard.
I started this subreddit with big ideas, but I never really got it off the ground. This falls under my usual motto, "I have a great idea, but I don't want to put a lot of effort into it."
That being said, you could write a script to convert all of your music and I could help you with that. I see it as a bit of a fun challenge, actually. What operating system are you running?
Edit: I'm also going to include this link. There are a ton of free LAME tools you can use on any operating system to transcode your 256kb songs to 192kb. Just make sure you are doing 192 VBR (variable bitrate) and not constant. This will allow the bitrate to drop lower when there's less data required and go a bit higher when there's more data required. http://lame.sourceforge.net/links.php#OpenSource
You are going to find that the variable settings in LAME encoding (or transcoding in this case) drop off significantly after -V4 and then pretty much plummet after -V5. http://wiki.hydrogenaudio.org/index.php?title=File:Lame-chart-2.png I would probably not go lower than -V4. This will be a setting in whatever program you ultimately use to convert your media.
If you can't get one of those apps do it all in bulk, let me know what operating system and I'll give you a batch file you can run from the command line to do it.
Edit2: Most scene releases are in -V2. Most audiophiles who tolerate MP3 use -V0.
Edit3: Make a backup of your music before you do this. If the conversion doesn't go well, you could destroy your audio library.
B.U.T.T. is a pretty easy, no-frills tool for recording and broadcasting.
You'll need the LAME dll for mp3 support, and you can record in CBR @ 320 kbps.
How to do this on a Linux box with a couple nifty command-line utilities:
- Download the youtube content using youtuble-dl (it's a nifty python script and it works well). ( ref. http://rg3.github.com/youtube-dl/ ):
$ youtube-dl url
- Convert the fetched youtube content (mp4 / flv file) into a WAV file using mplayer's mencoder(1) utility ( ref. http://www.mplayerhq.hu/ ):
$ mplayer -really-quiet -ao pcm:file="filename.wav" -novideo "filename.mp4"
- Use the lame(1) utility and libraries to convert into an MP3 encoded file ( ref. http://lame.sourceforge.net/ ):
$ lame --preset standard "filename.wav" "filename.mp3"
I'm sure there are plenty of other ways to do this (e.g., with FFMpeg, etc) but this is just one way that I'm happy with. Cheers.
The author uses ubuntu for his example but I can imagine it will work on windows too if you use another audio converter. Try this link for free audio converters http://lame.sourceforge.net/links.php#Windows
MP4 is an interesting choice for argument.
If people have to pay for your codec, they'll just switch to another one, regardless of how high quality / compressible your codec is. So if you want people to use your codec, you better use a fairly liberal license. In particular, very few people use DivX or Matroska, because, while they're free, they require specialized software as opposed to WAV/MP3, which every media program can handle, thanks to LAME.
Hi. Do you have a linux box or windows box?
Anyways, iTunes will do it for you if you go into the preferences and change the audio settings. Then one of the menus, like tools or something, will convert for you.
Or if you don't mind the command line, lame is awesome. Download it here: http://lame.sourceforge.net/using.php
A great example list here: http://lame.cvs.sourceforge.net/viewvc/lame/lame/USAGE
Quite simply, do: lame -V2 file.wav file.mp3
-V2 is a decent quality, -V0 is best variable quality.