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As you know, telephony involves voice transmission. Nobody needs a full bandwidth of 20Hz-20kHz for vocalise transmission system; up to 3.5kHz is enough for a clear distinguishable and identifiable voice. To be more precise, the speech absolute frequency band used in telephony is from 300 to 3400 Hz. When compressing into a common channel, for accurate assignation, protective frequency intervals along the edges are necessary, because the bandwidth is 4 kHz. When digitizing it turns out 8kHz. Now, in connection with the development of the thickness of communicating channels, the same Skype and others, boasting "increased" quality, use 16kHz, or equal 32kHz, which, however, is real not audible in a normal conversation (but IT is very distinctly distinguishable when worsened quality of the channel, merely when it worried the marketers).

So, almost all the sound files that are utilized in telephone are recorded with 8kHz digitization. To quicken up the processing of large flows, the condensation methods old are even as simple and aimed at a decent result when applied to the desired - speech compression. This is a simple digitization ( PCM ), plain delta codecs (ADPCM, G711 ), or tricky codecs ( GSM 06.10 ). These formats are "autochthonic" to telephone systems - star, freeswitch (and credibly others to a fault). In these formats, data is oven-ready for playacting by the system to people; in the same formats, the system can record records.

However, nowadays the entanglement is increasingly expanding around the planet, and people want to follow fit to listen to recordings of conversations, greetings, etc. on the network, where mp3 has become the "native" arrange ...

As a result, for the uncommon "listen to archive" function, the untrained solution is to configure the server to transcode the recordings from the call data formatting to MP3.

Everything would be fine, but:

• in mp3 recordings get along either big or worse;
• transcoding to mp3 requires load connected the waiter CPU;
• recoding occurs after the fact, and not happening the fly (although in that location is a cure for this to a fault);
• transcoded files are essentially solitary needful for the guest.

Seeing this disgrace, the soul of an engineer fell ill and began to requirement to do well. And it's not "to bed bad, but then how information technology was", namely, it's good and straightforward: in fact, the codecs used in telephone are designed for good results, and IT's exceedingly cut-rate. Thus why do an high-ticket encoding operation in MP3, then to do an expensive decoding operation from MP3 connected the client sensible because this decoder is already there? Let's just make this simplest decoder on the client, and that's it!

I was particularly surprised past the deficiency of these ready-successful decoders. This is how WavPlayer was born : a flash player for telephony files.

What can he do:

• Graphical user interface with a comic strip for jumping on a record, a GUI without information technology, loosely no UI
• API for managing and translation the entire interface on the JS side
• Codec support: PCM, G.711u / a, GSM 06.10, IMA ADPCM
• Indorse for formats: AU, WAV, several standard Open

And lately, users have added proxies to the orthodox MP3 histrion so that solely WavPlayer can cost misused to play both native and transcoded archives. (Initially, I did not practise this, forward that it was the JS side's concern to use any of the tatty-mp3 players, html5, or use WavPlayer).

Anyone who reads the descriptions of each of the codecs and formats will realize that the player is as simple as a traffic jam. Merely if that were so, it would have existed for a long time ... Therefore, I will briefly secernate the story of its creation.

Initially, only one was acknowledged to play sounds in a flashy: playing mp3 inserts. Whol. Nothing more. Starting with the 10th version, the sampleData event has appeared in the user interface flash.media.Wholesome, which allows generating and playing the generated sound. But as befits a flash, he does IT only in his own manner: only 44kHz, only stereophony, single 32bit floating-point numbers.

And we have 8kHz / 16kHz integers. If we just take the source information and just give out as-is, we buzz off something poorly readable and very high frequency. Conclusion? It is necessary to extrapolate the samples that we undergo - to neutralise some other dustup Resample.

When resampling, it is important to translate that flush with a simple doubling of the frequency, you can't just take and insert the "mean" numbers between the two samples - the resulting sound wish very "whistle" at high frequencies, because instead of a seamless sine curve we get a saw. The correct resampling is obtained by restoring the original smooth vocalise (with minimizing the 2d derivative), and Re-digitizing it at the desired frequency. This right smart we get the right smooth sound with the right sample plac.

Since I, of course, know the theory, only in practice I am very lazy, and the undertaking was to "play records" quite an acutely, it was necessary to solve quick. Flash I do not know, and a working automobile under Linux. I looked at the size of the flash compiler - for a hundred meters, it became so broken that I decided to find an alternative to rapidly and easily draw on a flash. Warm Googling gave a great choice - HaXe . A ovate C / java-look-alike language that privy be translated into respective target platforms, including the one I need - a flash. He was affected.

In the main, the first working layout was disorganized in haste :

There was a fogg externalise in which ogg files were manually decoded. From there, AudioSink was taken, which implements the press interface instead of pull: the cushion we write to, and when the trice wants the next piece of data, AudioSink sends it to it from the buffer. Non the most optimal and attractive effectuation, but prepare-made. As a resampler, Lanczos resampler implementation (the highest quality, supported sinc functions) from OpenJDK was taken forehead. The code is not the most optimal (later information technology was implemented on a pure Military action Script - it was able to accelerate almost 4 times), but it works (and I didn't involve anything other).
The interface is simple: draw a Triangulum when it is. On click, play () is launched and a square is drawn. By clicking, two rearing sticks are drawn.
For decoding G711, the code is taken from Sox, for PCM the code gave birth on its own.

And, of class, a spoonful of OOP in that barrel of tyrocode: File and Decipherer interfaces, which allow the important musician to abstract from a specific variation. Accurate, interfaces were born out needs, and not systematically, but when was it incompatible? File works like this - the input information of the data file is say, and shoved through the labor () method to the decoder. American Samoa soon as every headers are read, the file cabinet creates inside itself a decoder of the appropriate data format, and will begin to cram the audio information into it. The quick () method starts to return real, and from that moment on, completely strange stream metadata methods as wel become reasoned, and you can read the audio stream data using the getSamples () request, which wish return samplesAvailable () samples.

The operation of the decoder is also simple - information technology communicates the size of the sample in bytes so that the file can be cut into necessity packets to feed the decoder. The decoder is sequentially for converting the buffer information into unrivaled sample (into a signed float).

The of import problem that arises is the proper feeding of the sampler. Let me remind you that the resampler works on the principle of practical double conversion - supported the stimulus data with the stimulant sample distribution frequency, a smooth signal is restored, which is re-digitized at the output frequency. A signal is always needed to restore a indicate; therefore, first the decipherer must be fed with shut up of the desirable length, for initialisation. And throw out this silence from the first answer - then we will pose the correct resampling right from the beginning. In the unchanged way, after our data ends, the resampler must be Federal Reserve System hush up after - to get all the restored information.

And with such a macar our company of soldiers generates just how much data is needed at 44 kHz in the right form.

After the place player started working, it began to Be combed a little: first of all, support for more interlinking codecs, specifically gsm. It immediately became net that not all were decoded sample-by-bit, batch processing was needed here - so the decoder interface was redone for the input signal array + offset, yield lay out + offset, which returns how many samples were put to the output. To funding Overt files, most of the code is universal, it was moved to a separate general class, so that overriding the nominal - only the required parameters in the initializer. The GSM decoder itself was taken as was common where it was found, just changed quickly into the desired syntax. Oddly enough - it all worked out with a bang.

At the same time, the player ascertain interface was drawn from the JS code + the events of loading, playing, break were issued, allowing you to puff the state of the player in the browser as you want. The resulting cartesian product began to Be sawed into production. When they started testing, whatsoever problems got verboten, especially in the deeply idolized Id est, which the file squiffy in pieces seems to be 8k or 4k ... in the main, a ton of events was generated, I had to hewn the frequency of their generation.

Regrettably, information technology quickly became light up that no one had the desire to make an interface on JS. And so the decision was quickly and kneeled over aside gua inside. The musician began to mother internal events, and WavPlayerGui was created. His Mini heir remains as before - all button; plus, Full was created, which has the same button on the left and the progress bar on the right, showing the length, loudness of the loaded and the current position. Fortunate, that is, a bit more squares of pieces, the sizes of which changed in response to events.

A soon as this appeared, it became clear that in general it should also tick of on IT. In any case, listening to the recordings is only completely stupid when you pauperism to listen to the 3rd instant from the 15 minute ... You have to set assay (). The implementation of seek () in this case turned bent be the near unenviable task: since we possess no way to download the root file from an arbitrary position (we cannot vouch Scope support from the server, and IT is not thusly easy to do this in a flash), we had to circumscribe the possibilities of seek ( ) 'but only within the loaded part. But even in that case, we behave non stack away the full quantity of data transcoded at 44 kHz (memory, whimper, sorry), so if necessary to shift, the pursuit happens:

• We see to it to see if seek () is in the roam of ready 44kHz information - if thus, just make a set using the prepared data.
• if non, seek a sample opening from which playback should begin in terms of the master copy stream
• silence re-initializer,
• The stimulus watercourse is repositioned to the desired position,
• start playback.

Then in that respect were a hardly a cosmetic modifications from those World Health Organization started using it in the public, and over again there was a challenge - send away IMA ADPCM hold up be made. The format is sort o distasteful, from the point of view of beingness universal, it clothed that the information is not channel by duct, but mixed up in the same place, so I also had to transmit the decoded channel to the decoder; at the same prison term, I had to hit a trifle of universality for all other codecs, because the amount of output data, depending along the input for all others, is fixed and dolabriform; and here ... in general, it depends on - a clear history is required, and decoding cannot atomic number 4 started in any room from an arbitrary place. Accordingly, for attempt (), the function works like this:

• we bridle if seek () is in the range of ready 44kHz data - if so, just take in a sic victimisation the ready data
• if not, look for for a sample starting from which playback should begin in terms of the new current
• look for a sample where you can start decoding
• silence re-initializer,
• the input flow is repositioned to the decoding position
• do decryption and throwing to the position with which to start playacting
• start playback.

In gross, queerly sufficient, this too works. And at the moment, it is available for use by everyone: it does exactly what it needs, exactly the room IT should.
For a complete chill, it remains only to somehow make at last the very interface on JS that I assumed our web developers would do; plus to make a simple and comprehendible example of integration that you can put replicate-paste into your website, because most often this integration problem waterfall along the shoulders of the arrangement decision maker, and not the programmer ... So, to be continuing.

Project on Github | Online present.

DOWNLOAD HERE

GET WavPlayer - we are not looking for easy ways, we are paving them / Sudo Null IT News FREE

Posted by: salguerounnot1977.blogspot.com

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