Programs for changing the tonality by wt. Audacity is a free audio editor similar to SoundForge for Mac

◉ Create absolutely stunning audio recordings

MicroWave is an indispensable multitool for working with audio. Whenever you need to trim a few seconds off a music file, record your thoughts, split an audio file into many smaller ones, make a voice note, record an internet radio station* or digitize old tapes - MicroWave is there to help. It "s a tool as essential as a text editor. And because it" s that useful no Mac should miss it.

With MicroWave you will be able to:
digitize your beloved vinyl records and audio tapes
record your voice
record your favorite music from youtube/internet radio stations*
create engaging interviews
make great podcasts
optimize and improve the audio quality of those old charming vinyl records of yours
convert unfamiliar audio files so you can enjoy them on your iPod (ex. ogg/flac to mp3)
create your own unique and completely custom iPhone ringtones no one else has
come up with gripping audio effects for your iMovie productions
record captivating voice-overs for your slide shows
and much more ...

◉ What our customers say:

"Great Application. Brilliant Support." - Hendrik, Germany

"It is really easy to use, intuitive, quick and the sound quality is really good. No negatives." - Biff, United Kingdom

"A good audio editor, pretty powerful and easy to use" - Luis, Spain

"It did exactly what I wanted it to and only had to look up help once. Great value for money!" - Ned, Australia

"Simple, fast, elegant and right to the point." Rik, United States

◉ Detailed Feature List:
Editing:
Cut, Copy and Paste between multiple documents.
Persistent marker support with automatic silence detection.
Delete and trim audio.
Infinite and instantaneous Undo and Redo.
Undo and Redo states are saved with the document.
Supports importing of multiple audio files into one document.
Supports full screen editing.
Supports Stereo and Mono audio.
Support sampling rates 8khz - 96khz.
Supports bit depths 8bit, 16bit, 24bit, 32bit.
Supports conversion between all channel layouts, samples rates and bit depths.
Waveform zoom levels from 100% down to 1:1 sample resolution.
Supports editing of individual samples.

recording:
Supports recording from built in microphone and line-in.
Supports recording from any USB and FireWire Audio Device that is supported by OS X.
Supports recording from other applications (like browsers, etc) and internet streams*
Maximum recording length is only limited by free space on the hard drive.

Audio Effects:
fade in
fade out
Generate Silence
Silence Selection
Change Gain
NormalizeAudio
reverse audio
Delay (Echo)
Peak Limiter
Apply Distortion
Graphic Equalizer
low pass filter
high pass filter
Band Pass Filter
Multiband Compressor
Matrix Reverb
Change Pitch
Dynamics Processor
High Shelf Filter
Parametric Equalizer

Supported File Formats (import and export):
MP3 and OGG Vorbis
M4A, MP4, AAC and M4R (iPhone ringtones)
Flac Lossless
Apple Lossless (ALAC) encoded M4A files
WAV, AIFF and CAF
Can import audio tracks from MP4 and Quicktime videos (but can't save video files!)
Batch export: Splits document into multiple audio files based on markers.
Supports meta data (ID3 tags, etc) export and import

Support:
Detailed Manual included
24 hours support by email
We're excited about user feedback and feature suggestions :)

* Please note that recording audio from the internet or other applications requires installation of the free "Soundflower" driver - which doesn't take longer than 5 minutes. Detailed instructions can be found in the included manual.

Let's get back to the interface. A feature of Spark is the use of context menus in the program (yes, this is also possible on the Macintosh, despite the militant one-button function), in particular, this is used to add a module to a matrix cell. Each cell displays the name of the processing inserted into it and the level of the input signal. Unfortunately, only one window with processing parameters can be displayed at a time, to display the module interface, you need to double-click on the cell with it or "jump" directly by selecting the desired module from the Plug-In menu in the window for editing processing parameters.

Now about the sad. Firstly, the undo function is single-level (and this is in the 21st century!). Secondly, all editing operations are very slow: first, Spark saves a backup copy of the audio file, then it does the required operation (and this is also long, because instead of working with the edited piece, the program recalculates the entire file), then it calculates the waveform ( also the whole file). In general, extremely slow and inconvenient, especially with large (20-30 minutes) files. Apparently, such slow work is caused by the fact that during standard editing operations (delete, paste, etc.), Spark cannot work with part of the file, but only with the entire file, while creating a backup copy each time. Backup creation can, in principle, be disabled, but then you lose the undo altogether. Other programs seem more thoughtful to me in this place. There is another small drawback regarding the way the calculated waveform is stored. Most Macintosh programs do not store the calculated waveform in a separate file (like many Windows programs do), but in a special area of ​​the audio file called a "resource fork". That is, the user sees only one file, without any garbage of the same name that accumulates if you edit a file in several Windows editors. This is very convenient, especially since many programs "understand" each other and do not have to recalculate the waveform every time. Spark does not shine in this sense: not only does it not understand others, it also saves a graphical representation of the waveform in a separate file, next to the original one. A trifle, of course, but still unpleasant.

But enough about the sad stuff, let's move on to working with Spark samplers. Everything is very good here, the program can communicate with samplers both via MIDI and SCSI, and although the list of directly supported models is not as wide as that of Peak, you can import samples from Akai libraries from a computer CD-ROM directly into the program (and then already - and into the sampler).

I would especially like to note the means of restoring phonograms attached to Spark XL. These are two plugins: TC Declick and TC Denoise (unfortunately they are not VST and therefore can only work in Spark). Denoise (Fig. 9) works with the noise sample, subtracting it from the soundtrack, for Declick the sample is not required, the intensity and frequency of clicks are set by the controls. The quality of noise reduction in both modules is very good. Among the resource-dependent processing of this type, modules from Spark introduce the least amount of distortion into the soundtrack. I have not seen declickers comparable in quality to Declick, and perhaps only DINR from Digidesign can compete with Denoise. The latter has a lot of settings, and sometimes, "playing around" with the DINR parameters, you can achieve better results than with Denoise. In fairness, it should be noted that Denoise is quite easy to work with, and it produces good results even with default settings (however, if the result is not satisfactory, it is almost impossible to radically correct the situation, it remains to offer the module different noise samples).

If we talk about Spark in general, then I would like to note the good concept of the program and the intuitive interface. But the implementation of many functions of this editor is, unfortunately, not very good so far, so let's hope for improvements in future versions.

Testing
The abundance of functions of modern sound editors is impressive, but how honestly do programs perform these functions? It's no secret that many companies do not inform the user about the internal architecture and "pitfalls" of their software products. It happens, for example, that the bit depth of the processors built into the program can be low (16-bit), but at the same time the program processes 24-bit files without a twinge of conscience and without warning. The result is corrupted sound. In order to clarify some of the features of the programs that do not lie on the surface, a number of tests were carried out.

So, the first test is for the correctness of recording and playback. For it, I used a RME DIGI 96/8 Pro PCI card installed in a Power Macintosh 9600/350 computer. The board has digital interfaces in ADAT, SPDIF and AES/EBU formats. This board has Sound Manager drivers, but ASIO drivers were used for testing, as only 16-bit (or lower) audio can be played and recorded through the Sound Manager. The control system was Digidesign Pro Tools 442 with Pro Master 20 interface and Sound Designer II version 2.82 installed in Macintosh Centris 650. Despite its already quite respectable (by computer standards) age, this system still works perfectly and has been tested for correctness many times and in different parts of the world, so without hesitation it was chosen as the reference. So, the technique: the generated signal was taken (sampling frequency 44.1 kHz, bit depth 16 and 24 bits) with a sliding frequency (from 5 to 22000 Hz) lasting 30 seconds, played back by the Sound Designer program (via AES / EBU on Pro Master), and recorded in the programs under study (also via AES/EBU on the RME board); then, on the contrary, it was reproduced by programs, and recorded in Sound Designer. This used half-meter Apogee Wyde Eye wires with gold-plated Neutrik connectors. The resulting files in Sound Designer were cut off at the beginning and end of the signal (of course, accurate to the sample) and compared with the original generated file. For such a comparison, Sound Designer has a Compare Files command, in which the values ​​of each sample in one file are subtracted from the corresponding samples of the other, resulting in a sound file of the "difference" between the two files. If the two files match bit by bit (as it should with digital recording), then the resulting file will consist of all zeros. For complete certainty, the Find Peak command was used, and if the values ​​of all samples in the "differences" file are zero, then Sound Designer issues a message about this. If there are values ​​other than zero (respectively, the tested files do not match bit by bit), then the cursor will be placed on the loudest of such samples. To be sure, testing was carried out at least five times for each position, if the program recorded and played back correctly, then an additional test was conducted with a signal lasting five minutes.

The results of this test were rather disappointing. Of all the programs correctly (that is, the recorded file passed the null test), only Peak was able to record and play back, and even then only on 16-bit sound. All other null tests did not pass either on 16 bits or on 24 bits. Unfortunately, it is impossible to say exactly what exactly happens to the sound when recording and playing back programs, but it is obvious that this is not a 16- or 20-bit trunk (that is, simple rejection of "unnecessary" bits) - firstly, the distortions would be much more noticeable, and secondly, the bitscope of the SpectraFoo program, with which the results were analyzed (Fig. 10), showed the activity of all 24 bits of the signal. This is also not a dither, since the frequency of the difference signal increases as the frequency of the test signal increases. It is difficult to say unequivocally about the fault of the programs under study, since the system consisted of many elements (board, ASIO driver, program), although Peak showed that it is quite possible to work correctly with at least 16-bit sound with this board and driver provide. Various RME board driver settings have been used in an attempt to achieve correct operation, the sample rate master has changed (by default, the master is the playback device), but the results are always the same.

SpectraFoo bitscope readings on 16-bit (a) and 24-bit (b) signals.

Finally, let's talk about a number of program features identified in this test. Peak could not play the sound directly from the beginning of the file, the first few samples were "chewed", so to test this program, a file with a two-second silence at the beginning was used to allow Peak to "accelerate". After recording a 24-bit signal reproduced by the Spark program, it turned out that the recorded fragment of the test signal after cutting off the silence turned out to be four samples longer than the original one. What caused this is not known, but, apparently, the effect of this fact on the sound is insignificant.

Another study was conducted using the null test. A file was taken with the same test signal, but with a two-second silence before it (the same one used for the previous test of the Peak program). Then, in each program, a fade was made on this silence. Then all unnecessary silence was cut off, and the resulting file was compared with the original. This was done in order to find out whether the processing of the beginning of the file by the program affects the rest of the audio information. Spark and sonicWORX withstood this test with honor, but processing in the Peak program with a 24-bit file somehow affects the raw part. On the spectrum plot of the test signal (I used a tone of 1 kHz, more on this later), no distortion was visible, so it may not have such an effect. negative impact sound, but this phenomenon is still alarming.

The last null test was only done with the Spark program, because given function so far only she has. This function is to "assembly" a CD image from different audio files in the correct order. The disk image created by Spark (which can later be turned into an audio CD by the Adaptec Jam program) is a regular audio file with regions arranged to indicate the beginning and end of the track on the CD. Therefore, a 16-bit test file was taken, a three-track CD image was made from it, and then a null test was carried out. The result is that the original 16-bit files and the information contained in the disk image generated from them match bit by bit.

Then it was decided to test the quality of internal processing of audio editors. For this, a 24-bit file with a 1 kHz sinusoidal signal was taken and processed by VST modules in the following sequence: Waves Renaissance EQ, in which the equalization itself was completely turned off, and the level at the processing output was lowered by 3 dB, then the same module, but with output level increased by 3 dB, and at the end of the chain - Waves L1, which was used only for dithering from 32-bit VST processing to 24-bit output file (dither type 1, no noise shaping, all other L1 parameters did not change ). This was all converted into a new file, which was then analyzed by the spectrum analyzer of the Sonic Solutions workstation and the SpectraFoo program. IN this case all the programs studied showed their best performance - no nonlinear distortions were found on the spectrum graphs of the test files, which means that the VST system was built correctly and does not introduce any "gags" into the signal (Fig. 11).

Spectrum plot of a pure 24-bit 1 kHz sine wave:
a) - full spectrum in Sonic Solutions spectrum analyzer,
b) - spectrum above 1 kHz in SpectraFoo,
c) - for comparison, the same signal after trunking by 16 bits in the same scale as b,
d) - the same trunk, but on a different scale, allowing to consider the "forest" of emerging harmonics.

Lastly, the correctness of the built-in processing functions was tested using the same 1 kHz tone. The Change Gain command was applied to the test tone, first the level was lowered by 1 dB, then increased by the same 1 dB. In the same way, another of the most important processing was checked - fade. Then the test files were examined in the spectrum analyzer (Fig. 12). The results turned out to be quite good, that is, no non-linear distortions were found, which means that the built-in processing of all editors is true 24-bit. The only unpleasant exception is the fade of the Spark program, where an incomprehensible broadband noise appeared, which decreased along with the signal (accordingly, this is not a dither). In principle, the level of this noise is not high (maximum -110 dB), but its very appearance is alarming.

Graph of the spectrum of noise that occurs when fading in Spark:
a) - spectrum analysis in Sonic Solutions,
b) - in SpectraFoo,
c) - to compare fades in sonicWORX with the same resolution.

Thus, modern sound editors that work exclusively at the expense of the computer's central processor (and not at the expense of specialized DSPs, like Sound Designer or Sonic Solutions), despite their rather high functionality and speed, unfortunately, currently do not meet some basic requirements for professional work with sound. And although they have found their place in studio practice and can be successfully used to solve certain tasks, for any serious mastering (it is in this niche that some manufacturers position their products) these programs are not yet suitable.

conclusions
Which of the following editors to choose? There is no single answer, as always. And although all the described programs are located in approximately the same market sector, each has its own strengths and weak sides. If you prefer to quickly edit sound and work with effects in real time, then sonicWORX will suit this purpose as well as possible. And thanks to the included file processing modules, it will also come in handy for fans of non-trivial special effects. If you prefer to work not in real time (AP and AS formats) and often edit samples, then Peak will do. The latter is also not bad as a basic editor (cut, glue, etc.), although with simultaneous work With big amount files loses a little in speed to sonicWORX. Peak can also be recommended to the happy owners of Pro Tools TDM, as the only one of the described editors that allows you to use the DAE protocol for sound input / output. Pro Tools users will also be interested in Spark, but Direct Connect technology is not the best way to interact with Digidesign's brainchild. The main advantage of Spark, I think, is the ability to work with projects that can contain many separate audio fragments. For those who like to edit samples, Spark is also suitable - the unique function of importing from Akai format discs is very convenient, and the long time for calculating revisions can be neglected, for short fragments it almost does not poison life.

Table 1.
Common plug-in formats.

It's no secret that Mac OS X is rightfully considered the best platform not only for audio and video editing, but for creative work in general. And the point here is not so much in the technical capabilities of existing applications (although, to be honest, the possibilities are impressive), but in how convenient and practical the creative process itself is organized. That is why most designers, musicians and just creative people prefer Macs to the Windows platform. The topic of our today's review will be the Sound Studio audio editor, which allows you to easily record sound from various devices, as well as perform a wide range of audio editing actions.

When Sound Studio is launched, the user is immediately taken to the main program window. The application interface is quite simple and neat, the only criticism is the icons from the toolbar, the rough design of which does not fit in well with the look of the native mac program. The developers declared support for 11 localization languages, among which, unfortunately, there is no Russian. However, as we have already said, the interface is not so complicated that the lack of Russian localization caused any serious difficulties in performing basic operations.

Functionally, the window consists of several panels and two toolbars (control panel and toolbar), which are located at the top and bottom of the program window.

On main operating panel the spectrogram of the loaded or recorded audio material is displayed, and separately for each of its audio channels. Navigation bar is designed to quickly move to the desired section of the audio recording. This is quite convenient, especially if the scale of the spectrogram in the main window is increased and in order to get to the desired area, you have to use the slider.

Side panel is designed to display the names of markers placed on the spectrogram and their time coordinates. It is noteworthy that despite the lack of Russian localization, marker names can be written not only in Latin, but also in Cyrillic. To edit the name of a marker, double-click on it with the mouse.

Control Panel, located at the bottom of the window, contains tools to control the playback of the downloaded audio material and the recording of a new one. In addition to the block of control buttons, the panel has a pair of double indicators of the recording level and the volume level of the material being played.

Pro toolbar it is worth telling a little more, since funds for performing basic actions with the material have been withdrawn to it.

• Add Marker. A means of quickly placing a marker at the location of the track where the cursor is located. The markers themselves are displayed as vertical stripes, next to which their name is written. Not only the name of the marker can be changed (as we wrote a little higher), but also its location - just click on the marker and, holding the mouse button, drag it in the desired direction.

• normalize. Normalization of the entire track or its selected section. We click anywhere in the spectrogram of the track, and then, moving the cursor in the right direction, select the area for normalization, and then use the button normalize. If normalization is done immediately for the entire track, then nothing needs to be selected. In the next window, we set the level of normalization (from 0 to -24 dB), the method for searching for peak values ​​and calculating the desired changes, and then apply the settings.

• fade in. Gradually increase the sound volume. As a rule, it is set at the beginning of the track. It does not have any additional settings; to use it, it is enough to select the desired area and click on the button. Repeated pressing increases the effect of using the function, up to the complete disappearance of the sound.

• fade out. Gradual fading of sound volume. As a rule, it is set at the end of the track and is the exact opposite of the previous function, although it has a similar principle of use.
• fade special. Change the volume in a certain section of the track according to the settings set by the user. After selecting a section of the track and clicking on the desired button, the user enters separate window, in which he will have to build a graph of changes in volume. When creating a graph, you can immediately listen to the result using the playback buttons and save the graph as a preset in order to be able to use it to process other tracks in the future.

• Crop. The tool allows you to trim unnecessary material, leaving only what you need. In his work, he uses not markers, but selection with the mouse cursor. The principle of operation is simple: select the desired fragment and click on the tool icon - the program itself will delete everything that is not included in the selected area of ​​the track.
• Delete. And again we see the exact opposite with the same principle of using the tool. Function Delete allows you to remove everything that fell into the selected area of ​​the track without affecting the rest of the audio material. There are no additional settings for the tool.
• Split. An absolutely wonderful feature that allows you to cut a track into fragments with a given duration. The function works exclusively with the use of markers. We place them in the right places of the track, click on the function icon and get into a separate window in which you have to select the output format in which the fragments will be saved ( ), bitrate ( from 64 to 320 Kbps), specify the path and folder to save. If desired, you can fill in the main tags using the tag editor built into the program.

• Sample Rate. Ability to change the sample rate of the track in the range from 8000 to 192000 Hz. Setting the frequency is carried out by choosing one of the fixed values, or using a special wizard that allows you to visually see the change in file size and sound duration when setting one or another value.

• info. A fairly convenient tag editor, in appearance it is very similar to a similar tool in iTunes. Allows you to view and edit a wide range of tags ( tabs info, more info ), add lyrics ( Lyrics ) and album art ( Artwork ). All changes are made only in manual mode, automatic search and tagging is not provided.

In terms of importing, Sound Studio is quite omnivorous and supports a large number of audio formats, including common ones such as MP3, AIFF, AAC, FLAC and others. Export options are somewhat limited: only 12 formats are available to the user ( AAC, ADTS, AIFF, AIFF-C, Apple Lossless, CAF, FLAC, NeXT/Sun, Ogg Vorbis, Sound Designer II, Tab-delimited Text, WAV), and if the system has an installed LAME.Framework the program gets the ability to save files in the format MP3. When saving (exporting), the user has the opportunity not only to select the output file format, but also to set one or another encoding or sampling quality (depending on the format), and also to see the approximate file size that will result from the operation.

Sound Studio conditionally belongs to the class of applications in which only a part of the available material processing tools is displayed on the toolbar - a large part of the tools and functions are distributed over the tabs of the program menu.

• Edit

In this tab of the program menu, the user can use the function Silence, which allows you to completely remove the sound on a selected section of the track.

• Audio

Here we are interested in the possibility of setting up manual or automatic recording of sound from a microphone or other external sources connected to the Mac ( Audio->Auto Start/Stop Recording). The manual recording mode, when the user himself has to press the buttons to start recording or stop it, is quite simple and understandable and therefore does not require any explanation. But there are two automatic modes - recording and stopping by timer, or, most interestingly, depending on the signal level in the microphone or other source. For example, the program can be configured so that recording automatically starts at a sound level of 25 dB (a whisper of a person at a distance of 1 m from the microphone), pauses when the sound level drops to 15 dB (barely audible) and stays at this level for 3 seconds and finally stopped at 10 dB (almost inaudible) for 6 seconds.

It will also be useful Fourier spectrum analyzer, which does its calculations using fast Fourier transform algorithms ( Audio->Fourier Spectrum Analysis).

• Insert

Access to noise and tone generators. Here you can also add a few more to existing audio tracks, thereby creating multi-channel recordings. The number of tracks entirely depends on the processor power and volume random access memory computer.

• filter

The most interesting and sought-after program menu tab for most Sound Studio users. Numerous filters, of which more than a dozen are collected here, are grouped into 6 sections of the tab, according to the principle of operation and the nature of the changes made to the source audio material. Available filters include both your own and Apple Audio Units filters.

Among other things, the application boasts a developed system of shortcuts, which provides most of the functions and tools available in the program menu. Unfortunately, the shortcuts are hardcoded and the user will not be able to change them at will.

All Sound Studio settings menu options fit on one tab, where in three sections ( General, Audio, Colors) collected a few program settings.

Of the additional features of Sound Studio, I would like to note Ability to work in 32- and 64-bit modes. Fans of 64-bit should keep in mind that 32-bit plug-ins and filters of the program will work only if the application is launched in the same mode.

As a bonus, the developers provide the opportunity to download a set of Monbots programs from the site for free, which will automate the process of some operations and carry out batch processing of audio material. However, without Sound Studio installed, they will not be able to do much, since they use the functionality of the application for their work.

It will be convenient to use Sound Studio for digitizing analog audio recordings, creating podcasts, cutting ringtones and other operations related to sound processing. Subjective impressions from the work in the program are the most positive. In terms of functionality, and in some places and in appearance, Sound Studio resembles free app Audacity, which we already somehow on the pages of our site. By no means do we want to say that some developers copied the functionality of the program from others, but, nevertheless, a feeling of deja vu arises.

Sound Studio is distributed via Mac App Store and costs $30, and on the website of the developer company you can download a free trial version of the application, which will be fully functional during the first 15 launches.

• Audio editing features include cut, copy, paste, delete, silence, auto-trim, and more.
• Audio effects include sound amplification, normalization, equalizer, envelope, reverb, echo, reverse playback and many more.
• Built-in support for VST plugins gives professionals access to thousands of additional instruments and effects.
• Supports most audio file formats including mp3, wav, vox, gsm, wma, au, aif, flac, real audio, ogg, aac, m4a, mid, amr and many more.
• Batch processing allows you to apply effects and/or convert thousands of files in a single function.
• Search audio recordings and bookmark them for more precise editing.
• Create bookmarks and regions to make it easier to find, recall and collect segments of long audio files.
• Tools include spectrum analysis (FFT), speech synthesizer and voice changer.
• Sound restoration features include noise reduction and crackle removal.
• Supports sample rates from 6 to 96kHz, stereo or mono, 8, 16, 24 or 32 bits.
• Works directly with the MixPad multitrack audio mixer
• With an easy-to-use interface, you'll be editing in minutes

Company Audacity Developer Team decided to provide users with a modern and functional audio editor that will suit anyone, even a very demanding user. However, one of its biggest advantages is not even cross-platform, but free and rich functionality.

In contact with

So, Audacity can be safely put on a par with well-known paid solutions, and it will adequately withstand the competition. After installing the product on MacOS X it may seem that the program is a little rough and its design does not fit in with the platform, but the intuitive interface makes you forget about it in a matter of minutes. Moreover, the program supports 47 languages, including Russian (although not correct enough), which makes it easy to understand the controls.

A simple and intuitive interface is designed so that absolutely any user who does not have special knowledge in the field of sound processing can work with the program. The application allows you to work with files in various formats, including wav, aiff, au, mp3, wma, flac, etc. At the same time, to record a file in mp3 format, you need to additionally install the LAME encoder. To save files in wma, m4a, ac3 and amr formats, you need to download the FFmpeg library, which, like LAME, is not part of the audio editor due to licensing issues. Besides, Audacity will not be able to work with content protected by copyright.

To add a file to the editor, you can use the Finder functionality or simply drag and drop the desired track into the application window. After that, you can do almost anything with the audio file - cut, glue, change the tone, remove noise, and much more. At the same time, the program does not change the original file, using copies of the sections necessary for work. When saving, the changed parts are combined with the original file, however, only the changed parts can be saved separately. If a multi-channel audio file is saved, each channel can be saved separately.

In addition to processing digital content, the program allows you to digitize audio singles fed to the line input, as well as record from a microphone. At the same time, the start of recording can be automated when a certain signal volume is reached. Audacity offers the user great amount effects, as well as frequency analysis of individual parts of the audio file. With a thorough analysis of the program, you can find several shortcomings and errors in the work, but it can still quite easily compete with expensive analogues.