Modern Consumer Sound Appreciation

Over recent months, I have been racking my brain, trying to answer questions I have, about how sound that was compressed in the frequency-domain, may or may not be able to preserve phase-information. This does not mean that I, personally, can hear phase-information, nor that specific MP3 Files I have been listening to, would even be good examples of how well modern MP3s compress sound. I suspect that in order to stay in business, the developers of MP3 have in fact been improving their codec, so that when played back correctly, the quality of MP3s will stay in line with more-recent formats that exist, such as OGG Vorbis…

But I think that people under-appreciate my intellectual point of view.

For many months and years, I had my doubts, that MP3 Files can in fact encode ± 180⁰ phase-shifts, i.e. a stereo-difference channel that has the correct polarity with respect to the stereo-sum channel, over a range of frequencies. What my own musings have only taught me in recent days, is that in fact, MP3 is capable of ± 180⁰ phase-separation.

Further, similar types of compression should be capable of better phase-separation than that, If their bit-rates are set high enough, that not too many of their frequency-coefficients get chopped down – according to what I have reasoned out today.

What I also know, is that the sound-formats AC3 and AAC have as an explicit feature, to store surround-sound. MPEG-2 Video Files more-or-less require the use of the AC3 codec for sound, and MP4 Files absolutely require the use of the AAC codec. And, stored in its compressed format, the surround-effect only requires ± 180⁰ phase-accuracy.

This subject is orthogonal to debate which exists, about whether it is of benefit to human listeners, to have sound reproduced at very high sample-rates, or at great bit-depths. Furthermore, I do not fully know what good a very high sample-rate – such as “192kHz” – is supposed to do any listener, if his sound has been MP3-compressed. As far as I am concerned, ultra-high sample-rates have to do with lossless compression, or no compression, which also happen to produce the same file-sizes at that signal-format.

What I did was just check, in what format iTunes downloads music by default. And it downloads its music in AAC Format. All this does for me, is corroborate a claim a friend of mine made, that he can hear his music with full positioning, since that is also the main feature of AAC, and not of MP3.

Continue reading Modern Consumer Sound Appreciation

Some Thoughts on Surround Sound

The way I seem to understand modern 5.1 Surround Sound, there exists a complete stereo signal, which for the sake of legacy compatibility, is still played directly to the front-left and the front-right speaker. But what also happens, is that a third signal is picked up, which acts as the surround channel, in a way that neither favors the left nor the right asymmetrically.

I.e., if people were to try to record this surround channel as being a sideways-facing microphone component, by its nature its positive signal would either favor the left or the right channel, and this would not count as a correct surround-sound mike. In fact, such an arrangement can best be used to synthesize stereo, out of geometries which do not really favor two separate mikes, one for left and one for right.

But, a single, downward-facing, HQ mike would do as a provider of surround information.

If the task becomes, to carry out a stereo mix-down of a surround signal, this third channel is first phase-shifted 90 degrees, and then added differentially between the left and right channels, so that it will interfere least with stereo sound.

In the case where such a mixed-down, analog stereo signal needs to be decoded into multi-speaker surround again, the main component of “Pro Logic” does a balanced summation of the left and right channels, producing the center channel, but at the same time a subtraction is carried out, which is sent rearward.

The advantage which Pro Logic II has over I, is that this summation first adjusts the relative gain of both input channels, so that the front-center channel has zero correlation with the rearward surround information, which has presumably been recovered from the adjusted stereo as well.

Now, an astute reader will recognize, that if the surround-sound thus recovered, was ‘positive facing left’, its addition to the front-left signal will produce the rear-left signal favorably. But then the thought could come up, ‘How does this also derive a rear-right channel?’ The reason for which this question can arise, is the fact that a subtraction has taken place within the Pro Logic decoder, which is either positive when the left channel is more so, or positive when the right channel is more so.

(Edit 02/15/2017 : The less trivial answer to this question is, A convention might exist, by which the left stereo channel was always encoded as delayed 90 degrees, while the right could always be advanced, so that a subsequent 90 degree phase-shift when decoding the surround signal can bring it back to its original polarity, so that it can be mixed with the rear left and right speaker outputs again. The same could be achieved, if the standard stated, that the right stereo channel was always encoded as phase-delayed.

However, the obvious conclusion of that would be, that if the mixed-down signal was simply listened to as legacy stereo, it would seem strangely asymmetrical, which we can observe does not happen.

I believe that when decoding Pro Logic, the recovered Surround component is inverted when it is applied to one of the two Rear speakers. )

But what the reader may already have noticed, is that if he or she simply encodes his mixed-down stereo into an MP3 File, later attempts to use a Pro Logic decoder are for not, and that some better means must exist to encode surround-sound onto DVDs or otherwise, into compressed streams.

Well, because I have exhausted my search for any way to preserve the phase-accuracy, at least within highly-compressed streams, the only way in which this happens, which makes any sense to me, is if in addition to the ‘joint stereo’, which provides two channels, a 3rd channel was multiplexed into the compressed stream, which as before, has its own set of constraints, for compression and expansion. These constraints can again minimize the added bit-rate needed, let us say because the highest frequencies are not thought to contribute much to human directional hearing…

(Edit 02/15/2017 :

Now, if a computer decodes such a signal, and recognizes that its sound card is only in  stereo, the actual player-application may do a stereo mix-down as described above, in hopes that the user has a pro Logic II -capable speaker amp. But otherwise, if the software recognizes that it has 4.1 or 5.1 channels as output, it can do the reconstruction of the additional speaker-channels in software, better than Pro Logic I did it.

I think that the default behavior of the AC3 codec when decoding, if the output is only specified to consist of 2 channels, is to output legacy stereo only.

The approach that some software might take, is simply to put two stages in sequence: First, AC3 decoding with 6 output channels, Secondly, mixing down the resulting stereo in a standard way, such as with a fixed matrix. This might not be as good for movie-sound, but would be best for music.


 1.0   0.0
 0.0   1.0
 0.5   0.5
 0.5   0.5
+0.5  -0.5
-0.5  +0.5


If we expected our software to do the steering, then we might also expect, that software do the 90° phase-shift, in the time-domain, rather than in the frequency-domain. And this option is really not feasible in a real-time context.

The AC3 codec itself would need to be capable of 6-channel output. There is really no blind guarantee, that a 6-channel signal is communicated from the codec to the sound system, through an unknown player application... )

(Edit 02/15/2017 : One note which should be made on this subject, is that the type of matrix which I suggested above might work for Pro Logic decoding of the stereo, but that if it does, it will not be heard correctly on headphones.

The separate subject exists, of ‘Headphone Spacialization’, and I think this has become relevant in modern times.

A matrix approach to Headphone Spacialization would assume that the 4 elements of the output vector, are different from the ones above. For example, each of the crossed-over components might be subject to some fixed time-delay, which is based on the Inter-Aural Delay, after it is output from the matrix, instead of awaiting a phase-shift… )

(Edit 03/06/2017 : After much thought, I have come to the conclusion that there must exist two forms of the Surround channel, which are mutually-exclusive.

There can exist a differential form of the channel, which can be phase-shifted 90⁰ and added differentially to the stereo.

And there can exist a common-mode, non-differential form of it, which either correlates more with the Left stereo or with the Right stereo.

For analog Surround – aka Pro Logic – the differential form of the Surround channel would be used, as it would for compressed files.

But when an all-in-one surround-mike is implemented on a camcorder, this originally provides a common-mode Surround-channel. And then it would be up to the audio system of the camcorder, to provide steering, according to which this channel either correlates more with the front-left or the front-right. As a result of that, a differential surround channel can be derived. )

Continue reading Some Thoughts on Surround Sound