Category Archives: Audio

VLC Player starts left audio channel in mono

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VLC version 2.1.3 (latest) seems to start some MP3 files in stereo mode left and you’ll need to manually change it back to stereo every time a new file plays.

26-06-2014 15-30-49

There is no answer in the forums : link 1

Here is a solution to fix this annoyance.
There check the “All” under Show Settings (at the bottom left).
Go to Audio
Change “Stereo audio output mode” to Stereo ( The default is “unset” which seems to be the problem ).
Click SAVE to store the settings.

26-06-2014 15-29-02

Increase sound volume beyond 100% on Realtek HD audio

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Have you ever wanted to increase the volume of a video or music track, but found that everything was already set to 100%?

There are a few solutions that work well, but I found them to be either to cumbersome or not effective enough. Sometimes the “loudness” enhancement (build into windows) rather dampened the volume!! I also tried DSP and DX plugins, but they are a nightmare to implement on system sounds. The windows build in Audio Processing Objects, or APOs, (of which the loudness enhancement is one object) are a great feature, but I found nobody developing usable software that uses the library (??)

But if you are in luck of having Realtek HD Audio hardware, the driver software can be setup so it gives you an extra boost. And, as an added bonus, it does so without clipping.

Here is what you do:

Double click the “speaker” icon of the Realtek driver.
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Then configure your speakers as a 5.1 (or 7.1) surround set. (remove the check for the center, subwoofer and side pair. We don’t need them.)
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Enable Room Correction (!)

Then adjust the front left and front right speakers to +10 dB
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You will now have an extra 10 dB (which is a LOT) gain. This is especially noticeable when watching those youtube videos that have the audio mixed to low. Music will also gain from this trick, but when it is already loud, some compression will occur as to avoid clipping. Still the effect is much louder without extra hardware or intrusive (distorting equalizing) pay-software.

Have fun.

Android : Stereo recorder for Samsung Note 2

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I have been searching for a tool that can record in stereo on the Note 2 from Samsung. This seems to be a rare feature giving developers some difficulties to efficiently implement. Especially because the Note 2 uses two separate microphones for stereo recordings.

The Samsung Note 2 (and maybe also the S3, I’m not sure) has two microphones, one on the front and another on the back. If you record video, the source of the sound is comming from both these microphones, so giving you stereo sound. But most (if not all) of sound recording apps will allow you to select one of them, both not both. Even though they advertise to be able to record in stereo, they will not allow you to bind the front mic to the right channel and the back mic to the left.

Until I found Easy Voice Recorder. ( pro version ).


By using the “Sound input source” in the settings and selecting Camcorder microphone, the app will actually record in stereo using both microphones of your phone. The pro version gives you some extra features, worthwhile. You can choose bitrate and filetype, though mp3 is not supported — it uses AAC for compressed files.

Please post any comments or questions in the comment section below.

How to check quality of MP3 file

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I have been searching for a long time for tools checking the real quality of MP3 files. Many tools will just base their assumptions on the bitrate and encoder type alone (i.e. EncSpot). I think that’s rather insufficient. Though all knowledge on compression artifacts exists and computer programmer’s are out there making incredible stuff, all of the amazing audio analyzing on mpeg has to be done by ears…

Let’s see how we can at least do “something” to get a definite answer to the question: “Is this a bad file” without having to put our ears to the test. Mind though that it is in many ways like a cheap medical test where (in this case) a NO is 100% accurate, but a YES is just a way of saying: I’m not sure… let’s do more testing.

But first, an introduction.

With MP3 a revolution was born: Music became “portable”. But there’s a catch. MP3 is a lossy compression and will sacrifice quality for size. To do this without being noticed too much, MP3 uses a technique based on Psychoacoustics, where it filters out quite frequencies that are adjacent to loud frequencies. Scientists noticed that our ears mask those frequencies (we can’t hear them), so getting rid of these frequencies should go unnoticed. This way the encoder doesn’t have to “encode” those inaudible frequencies and thus can preserve a lot of space (10:1 is no exception). To preserve even more space (20:1) and so use lower bitrate, the encoder will filter out more frequencies. At some point that process becomes noticeable. That’s why very low bitrate MP3 sounds so horrible. ( This description is far from complete. ) If you want to know more about psychoacoustics, read this Wikipedia article..

MP3’s come in different qualities, i.e. how much compression is applied. Usually the compression correlates to the amount of kb/s (kilobits per second) is used to store and to reproduce the music. The more kb/s are available to store sounds, the better the quality can be, because the encoder can leave more of the quieter frequencies intact. And while this is “technically” sound, I’m going to prove to you that kb/s alone is far from a trustworthy indication of quality.

Digital MP3 music comes in different “qualities”, depending on how “lossy” the compression is. Let me list a few of them in order of “sounds horrible (like a cellphone)” to “sounds like the real thing”:

  1. 48 kb/s – This amount of compression sounds like your average AM portable radio and are mainly used to deliver speech.
  2. 96 kb/s – This sounds like a very poor quality Youtube video. This compression is often used to broadcast speech and music to mobile devices on GPRS (edge) or 3G, like your iPhone.
  3. 128 kb/s – This sounds acceptable to most people, but most (if not every) audiophile will notice the loss in quality without comparing it to the source. It’s is the most used compression for (private) MP3 music files on the internet and is mainly spread by filesharing, because it’s the best trade-of between size and quality.
  4. 192 kb/s – This sounds like CD quality to most people. Audiophiles will notice a slight degradation in quality, but only when comparing it to the source (so called A/B tests). It’s less portable, because it’s bigger and is better known as a better quality illegal download.
  5. 320 kb/s – This is the closest you can get to CD quality and still be compatible with most portable players. Audiophiles claim they can hear the difference, but it’s like tasting wine… (you know what I mean). This compression is used on many (legal) music download sites.
  6. 640 kb/s – Like twice as good as the best… (can that be?). Sadly some players won’t keep up with the amount of bits per second, or because the hard- and/or software won’t allow it.

Now, besides MP3 (which is MPEG 1 layer 3) there are many other lossy compression techniques out there, like mp2 (MPEG 2 layer 1 used by digital satellite), ogg, mwa and aac. AAC is known as the successor to MP3 and can achieve much better quality than at the same bitrate, through better use of the psychoacoustic schemes. An AAC file of 96 kb/s sounds like a 192kb/s MP3 file (debatable). AAC at 256 kb/s is widely used by iTunes and sounds awesome (very little trade-off between quality and compression, you get both).

Just for the record, there are some loss-less compression technique out there, like FLAC. These techniques will reduce the size of CD quality music without discarding any frequencies. The reproduction of this compressed file will be bit-by-bit identical to the source. Flac is mainly used for archiving CD collections and can achieve a considerable reduction in filesize.

After this lengthy, but still absolutely incomplete introduction, NOW ON TO THE MAIN DISCUSSION.

How to check the REAL quality

As we discussed above: 320 kb/s sounds better than 128 kb/s. Given this knowledge, some people make a habit of re-encoding bad 128 kb/s source material into 320 kb/s files (or even FLAC!!). In doing so, they incorrectly assume the quality will improve. It doesn’t, because once a music-file is compressed to 128 kb/s, the information removed to reduce size is lost forever. Nothing will ever recover what was lost. It’s like an image resized from 1000×1000 pixels to 100×100 pixels and then back to 1000×1000. The result will be a 90% loss of all information and the image will be blurry. There is no way anyone can sharpen it up again so that it contains the same details as the original 1000×1000 image.

Sadly, for whatever reason, there are 320 kb/s files out there that are in fact 128 kb/s MP3’s that have been upscaled somehow. But because 128 kb/s sounds pretty good as it is, it can be tricky to hear whether or not the file has been tampered with. There is (afaik) NO software out there that will do anything more then read the headers en look at the bits per second to tell you what’s good or not. According to audiophiles “listening” is the only way to tell if a file is good or bad, and that’s it. I will show you however, that a “bad” or “fraud” mp3 can be uncovered just by looking at it’s spectrum and without waking the neighbors.

The analysis is based on the frequency spectrum of the sound in the MP3 file. You can visualize this in many audio editors like Audition. We will be looking for giveaway features of most (if not all) MP3 encoders, i.e: the cut off frequency at 15 (or) 16 khz and clipping. Let’s look at some pictures:


The above image shows the spectral analysis of a CD quality file. The way it works is quite simple: The X axis is time, the Y axis is frequency and the illumination is the volume. Bright yellow means strong frequencies, red is weaker and black is none. This graph shows that all frequencies between 0 Hz en 20.000 Hz are very well present. Now, let’s look at a 128 kb/s compression of the same file:


Notice the difference in this 128 kb/s compressed file. All frequencies above 15,8 khz are gone (black). That’s because most MP3 encoders apply a “filter” on the sound before compressing it. Yes… This has absolutely nothing to do with psychoacoustics, but is just a rude filter that almost all encoders use. The cut-off frequency is thereby a strong indicator of the original bitrate. Most encoders cut off at 16 kHz for 128 kb/s to 8 kHz for 64 kb/s. Once these filters have been applied, the filtered-out frequencies are lost and cannot ever be recovered.

Note the small spikes that go up all the way to 22 khz. These are points where the music “clippes”. Clipping happens when the waveform is bigger that the container and is flattened at the top, it will produce strong harmonic distortion or overmodulation. Read more about clipping.


Above image: Example of clipping.

Once we saved the file again in MP3 320 kb/s the data previously lost in converting it to 128 kb/s is still missing. Consequently “upgrading” our file to 320 kb/s did not improve quality at all it only made the file almost 3 times as big. ( from 3,97 Mbyte to 9,93 Mbyte ).


Many so called CD quality 320 kb/s downloads are ripoffs. Let’s take a look at one of these wonderful 320 kb/s downloads that are in fact upgraded (in television midst they speak of upscaling) low quality MP3’s.


Now, this is quite an interesting one. Notice how the cutoff frequency seems to be there, but not quite “cutting off”… This file looks like it was encoded to 128 kb/s but with the cutoff filter disabled. As you can see the encoder tries to encode the high frequencies when they are really strong (hi hats + percussion), but eventually the encoder is loosing a lot of the information. Typically this will look like a largely blank upper frequencies area with brief spikes of high frequencies, strongly reduced in resolution.

I took the original to compare: The original (full bandwidth) file is very different from the (fake) 320 kb/s file. So what’s going on here?



Actually during my investigation I stumbled upon some annoying properties of Fraunhofer’s MP3 encoder. It seems that even with cutoff filter disabled (or set to 22kHz), the 320kb/s file (from the same original source as the image above) is somewhat filtered. There is a noticable “cutoff” line that isn’t present in the original file. Though the filtered 128 kb/s file will look flattened, this one still has some high frequencies in it. This file is bad, that’s for sure, but it’s not because of upscaling… !!!


This example above shows the same file, once in 320 kb/s unfiltered (left) and 128 kb/s unfiltered (right). Notice that their is some difference. The 320 file is probably better, but still not what we came to suspect from 320 kb/s files.


This is a comparison between the original and LAME (3.8). There is some degree of graining, but the overall frequency response seems better (wider) than FhG.

You can always spot a bad quality 320 kb/s MP3 by looking at the spectrum analyzer, but you can only be sure of the cause of the degradation when you see a very obvious 16khz (or less) cutoff. When the cutoff is overshooting, matters get more complicated. It can be a 128 kb/s source that was rendered without the filter, like we tried or it could be an original 320 kb/s file that was rendered with a bad encoder. In this case (to my surprise) the Fraunhofer encoder did not perform as expected, while the LAME encoder was quite good, although there is a little change in graininess between the original and the LAME 320kb/s encoded file.

Anyway. I hope I have been of any assistance doing these little tests today. 🙂