SST and codecs

Posted: Thu Aug 28, 2008 11:32 pm Post subject:

Indeed, the the (current) audio CD is such that the sample rate is of 44,100 samples per second. This means that the A/D converter (Analog to Digital) captures the sound 44,100 times in one second. Remember that when your favorite band/orchestra is playing, they are in the analog (real, concrete) world and that our beloved CDs are representants of the digital (virtual, 0s and 1s) world. Thus the need for conversion from one to the other.

Now, you can capture the sound 44,100 times per seconds, but the next step is to decide in how many bits you will put each sample captured. If you choose to do it with only one bit, you will get only 0 or 1 (2 combinations) at a time. However, if you decide to use 16 bits, you will get one of 65536 different variations in sound at a time. Standard CDs samples are stored in 16 bits each.

Good. We have 44,100 samples per second, each sample stored in 16 bits. But (again!) the standard CD stores its musics in a sterophonic way, i.e. in two distinct and independed channels. This is responsible by the spatial effect we gain when we listen to a stereo program.

Now we have all the elements. 44,100 samples per second, stored in 16bits each, and all this for each of two channels. Ready. We have 44,100 x 16 x 2 = 1411200 bits per second, or 1411Kbps, as you can see in your last list, audio rates, for CD Audio.

Now, about the lossless subject. The lossless is the sound recovered from the source without any kind of change. We use to say about the lossless music for that gotten from the CD in its "natural" state, or as a WAV file (44,100 x 16 x 2). This uses to be a large file.

There are two ways to change the sound in order to store it in less space than we would have if we would go to store it in its lossless form: a resample and a compression.

In the ressample, one transforms that 3 basic parameters of the music. For example, a CD music and its 44,100 x 16 x 2 would be resampled to 22,050 x 16 x 2, and then occupy half the space than the original. Or it would be resampled to 44,100 x 8 x 1, occuping 1/4 of the original size.

Yet the compression is different. It is an algorithm based on certain mathematical features of the music that allow us to "rewrite" the digital pattern according to this features. In a gross way, it is something like to check out the digital set of bits, searching for its parts that are not "well audible" by humans and then cutting them, completing the cut curve in order to not produce noise in the music. Is this process, a kind of resample occurs.

MP3, aacPlus, OOG and etc, are all about compression and, then their streams need to be decompressed (it is what the players do). All compressed sound has loss. Every one. What goes on is that some compressions are better done and then produce better sound quality when decompressed, but at the cost of bigger files and bigger times to compress.

I am not sure if the next paragraph is right. I am almost guessing in it. Hope that someone else can help us with it. Smile

Internet radio works with compressed streamings basically. And when you see that an aacPlus streaming is about 64k, this means that its compressed form is to be transmited in a rate of 64kbps in order to be reproduced accordingly, since the decompressor will do its task at this rate. The result of the decompressed sound will depend of the quality of the compression done.

Each process (MP3, aacPlus, etc) has its own algorithm and each one tends to be better to this or that sound characteristic. Personally, I like aacPlus very much.

And, finally, about the standards: they are settled accordingly with the technology available at the time they are to be decided. The CD standard was enough for a good time, but now, with the technology advance and the demand for more complex sound (countless channels for instance), a new standard is possible and even needed. Lets wait what is coming on! Smile

Well, hope have help you somehow. I have to sleep! Night!

And welcome to SST forums! Smile

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