The most common misconceptions about digital audio. Complete guide to settings in OBS Studio

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> How to change MP3 bitrate?

Introduction.

MP3 is the most common audio format, supported by most modern players. Files in this format are small in size, but when encoding into MP3, loss of sound quality is inevitable. The quality in this format can be characterized by bitrate. Bitrate is a parameter indicating how much data is allocated to a certain segment of audio. The bitrate is set during encoding and can only be changed through recoding.

If you increase the MP3 bitrate, the sound quality will only get worse. But if you make a smaller bitrate out of a larger one, then the gain will be in the size of the music file. If you are looking for a program to change bitrate And reduce MP3 size, then AudioConverter Studio is ideal for this.

Step one: Download and install the program.

Download AudioConverter Studio to your chosen folder and run the installation. Follow the installation instructions to complete the process.

Step two: Launch the program. Selecting MP3 files for conversion.

Run MP3 file converter. A wizard will open, but for our case we will use in the usual way work with the program and close it.

Now you need to add MP3 files for which you want to change the bitrate to the program. Click the Browse button to the right of address bar to go to the desired folder. Immediately after this, the music files will appear in the list.

Step three: Setting the MP3 bitrate and other parameters.

On the right side of the program window two sections are visible. Let's look at the "Output Format" section.

Go to the MP3 tab and select required settings audio quality.

In the "Tags and file name format" section, set "Old filename". This means that the program will name the new files the same way as the old ones were named. For ID3 tags, we will select the "Copy existing" option to copy the tags to the new MP3s.

Step Four: Selecting the Output Folder. Changing the bitrate of MP3 files.

So, all preparations have been completed.

On the toolbar, click "Add All" to add all MP3s in the current folder to the conversion list. In the "Output Files" section, specify the folder where you want to save the converted audio. Please note that in this case you cannot use the folder with the source files.

Just below, in the table, information on future MP3s is displayed.

Click "Convert" on the toolbar to change bitrate of MP3 files.

Bitrate (from English. bitrate) is the number of bits (units of information) used to store one second of video or audio recording. The most common unit of measurement for bitrate is kilobits per second (Kbps). There are two types of bitrates most commonly used: constant and variable. Constant bitrate does not change throughout the entire file, while variable bitrate changes depending on the richness of the audio or video.

Audio and video bitrate is one of key characteristics multimedia files, affecting their quality and size. The higher the bitrate the music or video was recorded, the better their quality will be and the larger the recording files will be.

Accordingly, changing the bitrate in one direction or another can increase or decrease the file size. But with the impact on the quality of recordings, everything is a little more complicated. While decreasing the bitrate naturally leads to a deterioration in the quality of the source file, the opposite operation does not affect the quality in any way. Even if you set the maximum bitrate for a song or movie, the audio and video quality of your file will remain the same.

As you can see, there is no particular point in increasing the recording bitrate: as a result, you will get a larger file with the same quality. But it is very possible to reduce the bitrate in order to reduce the recording size. Want to try changing the bitrate of your songs or movies? Download Movavi Converter Video is a convenient utility with which you can easily change the bitrate of video and audio recordings, be it files in popular formats MP3, WMA, AVI and MP4 or recordings placed in more exotic containers.

This guide is dedicated to the program Open Broadcaster Software e (hereinafter OBS) and its settings for streaming on Twitch.tv And Cybergame.tv. So let's start in order.
1. First you need the program itself OBS- to do this, go to the website http://obsproject.com/ and go to the section Download and download the distribution. We install it following the installer's instructions.
2. Launch the program. And we’ll make settings for the stream on Twitch.tv
2.1. Next we need to go to the program settings - Settings -> Settings

2.2. In the menu that appears, we can change the Language, we can also immediately name our profile (Profiles are some kind of settings presets, for example, you can create a profile for streaming on Twitch in 720p quality, and create a profile for streaming on Cybergame in 1080p, and switch between them with just a couple of mouse clicks). First, let's create our first profile. To do this, you need to click in the window to the right of the inscription " Profile:" erase everything that is written there and write your name, for example I will write "720p Twitch", and press the button Add.

Let’s also immediately look at the steps required to delete a profile. When you install the program, a profile is automatically created for you " Untitled", now we will delete it with you. To do this, to the right of the line " Profile:"there is a down arrow (drop-down menu) select a profile there" Untitled" and press the button " Delete".

2.3. Go to the tab " Coding". This window displays some of the most important settings for your stream, in most cases the picture quality during dynamic scenes will depend on them.
From September 1st Twitch.tv began to require streamers to set a Constant bitrate, accordingly we put a checkmark next to CBR (constant bitrate) We also check for the presence of a daw CBR padding(if absent, put it in!).
To stream on Twitch.tv with permission 1280x720 I would advise using a bitrate in the range of 2000-2500 (at 2000 the picture will be less clear, but fewer viewers will complain about friezes; at 2500, on the contrary, the picture will be of higher quality, but viewers may begin to complain about more frequent friezes in the picture). For example, let's take something in between - 2200
Below we see Audio settings, everything is simple here, set Codec: AAC And Bitrate: 128.

2.4. Broadcast. In this tab we must select the broadcast service and indicate the channel key in it. In our case it will be Twitch.tv. So we set:
Mode: Live
Broadcasting service: Twitch / Justin.tv
Server: EU:London, UK(you can have another one starting with EU:)
Play Path/Stream Key (if available): here we must insert the key to our channel. To get it you need to go to the Twitch website, create an account/log in and follow the following link http://ru.twitch.tv/broadcast On the right you will see a button " Show Key"

click on it and copy the key that appears (starts with live_). Be VERY careful and copy the ENTIRE key, a mistake in 1 character will not allow you to start the stream.
Auto reconnect: Checkmark
Auto reconnection delay: 10(possibly less given number determines how many seconds after the stream crashes OBS will try to start it again.)
Delay (sec): 0(As a rule, the delay is set on the stream of Company or Special battles, the delay is set in seconds, for example, to set the delay in 10 minutes need to write 600 )

Please note that OBS writes in red, this is precisely due to the new requirements Twitch.tv which came into force on 09/01/2013. (we will fix this below)
2.5. Tab Video. Here we select the resolution in which viewers will see our picture. IN Base Resolution: select Custom: and enter 1280 and 720.
Frames per second (FPS): put 30

2.6. Audio. Microphone and sound settings in general. Choose playback device sound (usually Speakers) we also choose Microphone if you want to use the Push To Talk system (so that what you say can be heard on the stream only when you press a certain button), then check the box next to Use Push to Talk and to the right, select the window and press the button to which we want to assign this function (for example, I assigned it to Q)
NiG Latency (ms): 200(if viewers complain that the endings of your phrases often disappear, then you can increase this value (but do not overdo it, I advise you to increase it by 200 and carry out tests. Personally, everything is fine with a value of 200)
Hotkey Microphone On/Off And Hot key Enable/Disable sound- you can set hotkeys for these actions (they will mute the microphone and sound on the stream)
Application gain (multiplier): 1(this setting increases the sound of all applications, I advise you to leave it at 1, but if suddenly by setting the sound in the game to maximum, viewers complain that they cannot hear the sound, you can change given value(I recommend adding 1 at a time) (I’m fine even with a value of 1)
Microphone Gain (Multiplier): 1(this setting increases the sound of the microphone, I advise you to leave it at 1, but if suddenly by turning up the microphone volume, the audience complains that they can’t hear you, you can change this value (I recommend adding 1 at a time) (I’m fine with the value 1)

2.7. Advanced tab.
Multi-threaded optimization checkbox
Process priority Medium
Scene buffering time (ms): 400
Preset x264 CPU: Veryfast(for owners of super-powerful processors you can install faster or fast, I don’t recommend it, because... the load on the CPU will increase greatly)
Interval key frames(sec, 0=auto): 2(Twitch requirement)
CFR (Constant Frame Rate) check mark
Adjust sound to video timing checkbox(there is a rare bug that the sound lags behind the video and this checkbox fixes it, one of our streamers encountered this)

3. Settings for Cybergame.tv
3.1. Create a profile - to do this, go to the tab Are common. to the right of Profile: write the profile name, for example: " 1080p Cybergame" and click Add.

Note! If you have selected a profile (for example, 720p Twitch) and you create a new one, then it will completely copy all the settings of the previous profile, and you will only need to adjust it a little.

3.2. Coding. To stream on Cybergame.tv it is not necessary to use CBR (constant bitrate) but we still use it, because We use restreaming on Twitch.tv.
Maximum Bitrate (Kbps): 3700(For 1080p stream on Cybergame.tv I recommend using bitrate 3500-4000 (since the service Cybergame.tv broadcast servers are located in Russia(y Twitch.tv coming in Europe) then the bitrate can be set higher, for example, if you make a 720p stream on Twitch, you use a bitrate of 2000-2500, then for the same stream on Cybergame.tv you can use a bitrate of 2500-3000))
Audio: AAC - 128

3.3. Broadcast
Mode: Live
Broadcast service: Custom
Server: In order to find out the server, you need to log in/register on the Cybergame.tv website - go to your account using the link http://cybergame.tv/cabinet.php, select the "Channel" tab and copy what is next to Broadcast settings:(For example rtmp://st.cybergame.tv:1953/live)
Play Path/Stream Key (if available): And here we copy what is next to Stream Name (Path):(but first you need to click the Display button so that numerous stars disappear) usually starts with your nickname. (copy from the same page from which the Server was copied)

3.4. Video
because We plan to stream in 1080p, then we write in Custom: 1920 1080
Frames per second (FPS): 30

3.5. Settings Audio And Expanded you can take exactly the same ones as for streaming on Twitch.tv.

4. Settings scenes And Sources
First, let's figure out what a Scene is and what a Source is.
A scene is a profile that contains one or more source(s). Those. for convenience, we can create scenes with the name of the games: "WoT" "WoWP" "CS", etc. and each scene will have its own sources configured, for example, in the “WoT” scene there will be a source with game capture, a source with your webcam, etc. those. Sources are some kind of layers, and the source that is higher in the list will be in the foreground, and the one that is lower will be in the background. Well let's get down to business.
4.1. Initially we have Scene let's rename it to "WOT" To do this, right-click on it and select "Rename"

we write "WOT" click ok. we get a scene with a title WOT
4.2. Next, let's add to this scene source with a picture of the game. To do this, the game must be running!
Right-click in an empty window Sources: and choose Add -> A game

Enter a name, for example WOT.
We have a window. IN Application: we should find our game in the drop-down menu : WoT Client
also check the box "Stretch image to full screen" And "Mouse Capture" click OK

Also in the sources you can add Slide show(several pictures changing periodically) Image(static picture or gif animation) Text(any text) Device(Webcam).
You can view the result of the picture by clicking on the button "Preview"

You will have a video with your layers. As I wrote above, the source that is higher is in the foreground, and the one that is lower is in the background. If you plan to overlay pictures/text on top of the game, then the game should be at the very bottom of the list of Sources.

In order to adjust a particular layer (its size or position on the screen) - WITHOUT leaving the Preview mode, click on Scene change and click on the source you want to edit. A red frame will appear around the selected source, by stretching it you can change the size of the source itself. You can also move the source to any location.

We also see red “bars” that will help you adjust the volume balance between the microphone and other sounds (I’m not your advisor here, this is very individual and needs to be agreed with the audience.)

Well, the finish line, in order to start the broadcast, stop the preview and click Start broadcast.

It is very important that when you stream you don’t have Personnel loss. If you have frame loss, then perhaps you have problems with the Internet or you simply do not have enough of your channel for the current stream settings. Try reducing the bitrate.

Guide prepared neRRReQuCb especially for ACES TV viewers.

Debunking popular myths about digital audio.

2017-10-01T15:27

Audiophile's Software

Note: For better understanding In the text below, I highly recommend that you familiarize yourself with the basics of digital audio.

Also, many of the points raised below are covered in my publication “Once again about the sad truth: where does good sound actually come from?” .

The higher the bitrate, the better the quality of the track.

This is not always the case. First, let me remind you what bitray is T(bitrate, not bitraid). This is actually the data rate in kilobits per second during playback. That is, if we take the size of a track in kilobits and divide it by its duration in seconds, we get its bitrate - the so-called. file-based bitrate (FBR), usually it is not too different from the bitrate of the audio stream (the reason for the differences is the presence of metadata in the track - tags, “embedded” images, etc.).

Now let's take an example: the bitrate of uncompressed PCM audio recorded on a regular Audio CD is calculated as follows: 2 (channels) × 16 (bits per sample) × 44100 (samples per second) = 1411200 (bps) = 1411.2 kbps . Now let’s take and compress the track with any lossless codec (“lossless” - “lossless”, i.e. one that does not lead to the loss of any information), for example the FLAC codec. As a result, we will get a bitrate lower than the original one, but the quality will remain unchanged - here is your first refutation.

There is one more thing worth adding here. The output bitrate with lossless compression can be very different (but, as a rule, it is less than that of uncompressed audio) - this depends on the complexity of the compressed signal, or more precisely on data redundancy. Thus, more simple signals will be compressed better (i.e. we have a smaller file size with the same duration => lower bitrate), and more complex ones will be worse. This is why lossless classical music has a lower bitrate than, say, rock. But it must be emphasized that the bitrate here is in no way an indicator of the quality of the audio material.

Now let's talk about lossy compression. First of all, you need to understand that there are many different encoders and formats, and even within the same format, the encoding quality of different encoders may differ (for example, QuickTime AAC encodes much better than the outdated FAAC), not to mention the superiority of modern formats (OGG Vorbis, AAC , Opus) over MP3. Simply put, of two identical tracks encoded by different encoders with the same bitrate, one will sound better and another will sound worse.

In addition, there is such a thing as upenvelope. That is, you can take a track in MP3 format with a bitrate of 96 kbps and convert it to MP3 320 kbps. Not only will the quality not improve (after all, the data lost during the previous 96 kbit/s encoding cannot be returned), it will even worsen. It’s worth pointing out here that at each stage of lossy encoding (with any bitrate and any encoder), a certain amount of distortion is introduced into the audio.

And even more. There is one more nuance. If, say, the bitrate of an audio stream is 320 kbps, this does not mean that all 320 kbps were spent on encoding that very second. This is typical for encoding with a constant bitrate and for those cases when a person, hoping to get maximum quality, forces the constant bitrate to be too high (for example, setting 512 kbps CBR for Nero AAC). As is known, the number of bits allocated to a particular frame is regulated by a psychoacoustic model. But in the case when the allocated amount is much lower than the set bitrate, even the bit reservoir does not save (read about the terms in the article “What are CBR, ABR, VBR?”) - as a result, we get useless “zero bits” that simply “finish off” » frame size to the desired size (i.e., increase the flow size to the specified one). By the way, this is easy to check - compress the resulting file with an archiver (preferably 7z) and look at the compression ratio - the higher it is, the more zero bits (since they lead to redundancy), the more wasted space.

Lossy codecs (MP3 and others) are able to cope with modern electronic music, but are not capable of high-quality encoding of classical (academic), live, instrumental music

The “irony of fate” here is that in fact everything is exactly the opposite. As is known, academic music in the vast majority of cases follows melodic and harmonic principles, as well as instrumental composition. From a mathematical point of view, this results in a relatively simple harmonic composition of music. Thus, the predominance of consonances produces a smaller number of secondary harmonics: for example, for a fifth (an interval in which the fundamental frequencies of two sounds differ by one and a half times), every second harmonic will be common to the two sounds, for a fourth, where the frequencies differ by one third - every third, and etc. In addition, the presence of fixed frequency ratios, due to the use of equal temperament, also simplifies the spectral composition of classical music. The live instrumental composition of the classics determines the absence of noise characteristic of electronic music, distortion, sharp jumps in amplitude, as well as the absence of an excess of high-frequency components.

The factors listed above lead to the fact that classical music is much easier to compress, first of all, purely mathematically. If you remember, mathematical compression works by eliminating redundancy (describing similar pieces of information using fewer bits) and also by making predictions (aka. predictors predict the behavior of the signal and then only the deviation is encoded real signal from the predicted one - the more accurately they match, the fewer bits are needed for encoding). In this case, the relatively simple spectral composition and harmony lead to high redundancy, the elimination of which provides a significant degree of compression, and a small number of bursts and noise components (which are random and unpredictable signals) determines good mathematical predictability of the vast majority of information. And I’m not even talking about the relatively low average volume of classical tracks and the frequent intervals of silence, for which practically no information is required to encode. As a result, we can losslessly compress, for example, some solo instrumental music to bitrates below 320 kbps (TAK and OFR encoders are quite capable of this).

So, firstly, the fact is that the mathematical compression underlying lossless encoding is also one of the stages of lossy encoding (read Understanding MP3 encoding). And secondly, since lossy uses the Fourier transform (decomposition of the signal into harmonics), the simplicity of the spectral composition even makes the encoder’s work doubly easier. As a result, comparing the original and encoded classical music samples in a blind test, we are surprised to find that we cannot find any differences, even at a relatively low bitrate. And the funny thing is that when we start to completely reduce the encoding bitrate, the first thing that reveals differences is background noise in recording.

As for electronic music, coders have a very difficult time with it: noise components have minimal redundancy, and together with sharp jumps (some kind of sawtooth pulses) they are extremely unpredictable signals (for coders who are “tailored” to natural sounds that behave completely otherwise), the direct and inverse Fourier transform with the rejection of individual harmonics by the psychoacoustic model inevitably produces pre- and post-echo effects, the audibility of which is not always easy for the encoder to assess... Add to this high level RF components - and you get a large number of killer samples, which even the most advanced encoders cannot cope with at medium-low bitrates, oddly enough, especially among electronic music.

Also amusing are the opinions of “experienced listeners” and musicians who, with a complete lack of understanding of the principles of lossy coding, begin to claim that they hear how the instruments in music after coding begin to go out of tune, the frequencies float, etc. This might still be true for antediluvian cassette players with detonation, but in digital audio everything is accurate: the frequency component either remains or is discarded, there is simply no need to shift the tonality. Moreover: the presence of a person’s ear for music does not at all mean that he has good frequency hearing (for example, the ability to perceive frequencies >16 kHz, which disappears with age) and does not at all make it easier for him to search for lossy coding artifacts, since distortion These have a very specific character and require the experience of blind comparison of lossy audio - you need to know what and where to look.

DVD-Audio sounds better than Audio CD (24 bits vs. 16, 96 kHz vs. 44.1, etc.)

Unfortunately, people usually look only at numbers and very rarely think about the impact of a particular parameter on objective quality.

Let's first consider the bit depth. This parameter is responsible for nothing more than the dynamic range, i.e., the difference between the quietest and loudest sounds (in dB). In digital audio, the maximum level is 0 dBFS (FS - full scale), and the minimum is limited by the noise level, i.e., in fact, the dynamic range in absolute value is equal to the noise level. For 16-bit audio, the dynamic range is calculated as 20 × log 10 2 16, which equals 96.33 vB. At the same time, the dynamic range of a symphony orchestra is up to 75 dB (mostly about 40-50 dB).

Now let's imagine real conditions. The noise level in the room is about 40 dB (do not forget that dB is a relative value. In this case, the hearing threshold is taken as 0 dB), maximum volume music reaches 110 dB (so that there is no discomfort) - we get a difference of 70 dB. Thus, it turns out that a dynamic range of more than 70 dB in this case is simply useless. That is, at a higher range, either loud sounds will reach the pain threshold, or quiet sounds will be absorbed by surrounding noise. It is very difficult to achieve an ambient noise level of less than 15 dB (since the volume of human breathing and other noise caused by human physiology is at this level), as a result, a range of 95 dB for listening to music turns out to be completely sufficient.

Now about the sampling frequency (sampling frequency, sample rate). This parameter controls the time sampling frequency and directly affects the maximum signal frequency that can be described by a given audio representation. According to Kotelnikov's theorem, it is equal to half the sampling frequency. That is, for the usual sampling frequency of 44100 Hz, the maximum frequency of the signal components is 22050 Hz. The maximum frequency. which is perceived by the human ear is slightly above 20,000 Hz (and then at birth; as we grow older, the threshold drops to 16,000 Hz).

This topic is best covered in the article Downloads in 24/192 format - why they don’t make sense.

Different software players sound differently (e.g. foobar2000 is better than Winamp, etc.)

To understand why this is not the case, you need to understand what a software player is. Essentially this is a decoder, handlers (optional), an output plugin (to one of the interfaces: ASIO, DirectSound, WASAPI. etc.), and of course GUI ( GUI user). Since the decoder in 99.9% of cases works according to a standard algorithm, and the output plugin is just part of the program that transmits the stream to the sound card through one of the interfaces, the only reason for the differences can be the handlers. But the fact is that handlers are usually disabled by default (or should be disabled, since the main thing for good player- be able to convey sound in its “pristine” form). As a result, the only subject of comparison here can be possibilities processing and output, which, by the way, are very often not necessary at all. But even if there is such a need, then this is a comparison of processors, and not of players.

Different driver versions sound different

This statement is based on banal ignorance of the principles of operation of a sound card. The driver is software, necessary for effective interaction of the device with operating system, which also usually provides a graphical user interface for the ability to manage the device, its parameters, etc. The sound card driver ensures that the sound card is recognized as sound device Windows informs the OS about the formats supported by the card, ensures transmission of an uncompressed PCM (in most cases) stream to the card, and also gives access to settings. In addition, if there is software processing (using CPU tools), the driver may contain various DSPs (processors). Therefore, firstly, with effects and processing disabled, if the driver does not provide accurate PCM transmission to the card, this is considered a grave mistake, a critical bug. And this happens rarely. On the other hand, differences between drivers may be in updating processing algorithms (resamplers, effects), although this also does not happen often. Moreover, to achieve highest quality effects and any driver processing should still be excluded.

Thus, driver updates are mainly focused on improving stability and eliminating processing errors. Neither one nor the other in our case affects the quality of playback, therefore in 999 cases out of 1000 the driver has no effect on the sound.

Licensed Audio CDs sound better than their copies

If no (fatal) read/write errors occurred during copying and optical drive device on which the copy disc will be played, there are no problems with reading it, then such a statement is erroneous and easily refuted.

Stereo encoding mode gives better quality than Joint Stereo

This misconception mainly concerns LAME MP3, since all modern encoders (AAC, Vorbis, Musepack) use only Joint Stereo mode (and this already says something)

To begin with, it is worth mentioning that the Joint Stereo mode is successfully used with lossless compression. Its essence lies in the fact that before encoding the signal is decomposed into the sum of the right and left channels (Mid) and their difference (Side), and then separate encoding of these signals occurs. In the limit (for the same information in the right and left channels), double data savings are obtained. And since in most music the information in the right and left channels is quite similar, this method turns out to be very effective and allows you to significantly increase the compression ratio.

In lossy the principle is the same. But here, in the constant bitrate mode, the quality of fragments with similar information in two channels will increase (in the limit, double), and for the VBR mode in such places the bitrate will simply decrease (do not forget that the main task VBR mode - stably maintain a given encoding quality using the lowest possible bitrate). Since during lossy encoding, priority (when distributing bits) is given to the sum of channels, in order to avoid deterioration of the stereo panorama, dynamic switching between Joint Stereo (Mid/Side) and regular (Left/Right) frame-based stereo modes is used. By the way, the reason for this misconception was the imperfection of the switching algorithm in older versions of LAME, as well as the presence of the Forced Joint mode, in which there is no automatic switching. IN latest versions LAME Joint mode is enabled by default and it is not recommended to change it.

The wider the spectrum, the better the quality of the recording (about spectrograms, auCDtect and frequency range)

Nowadays, on forums, unfortunately, it is very common to measure the quality of a track “with a ruler using a spectrogram.” Obviously, due to the simplicity of this method. But, as practice shows, in reality everything is much more complicated.

And here's the thing. The spectrogram visually demonstrates the distribution of signal power over frequencies, but cannot give a complete picture of the sound of the recording, the presence of distortions and compression artifacts in it. That is, essentially all that can be determined from the spectrogram is the frequency range (and partly the spectrum density in the HF region). That is, in the best case, by analyzing the spectrogram, an upconvert can be identified. Comparing spectrograms of tracks obtained by encoding with different encoders with the original is complete absurdity. Yes, you can identify differences in the spectrum, but determining whether (and to what extent) they will be perceived by the human ear is almost impossible. We must not forget that the task of lossy coding is to ensure an indistinguishable result human ear from the original (not by eye).

The same applies to assessing the quality of encoding by analyzing the output tracks with the auCDtect program (Audiochecker, auCDtect Task Manager, Tau Analyzer, fooCDtect - these are just shells for a one-of-a-kind console program auCDtect). The auCDtect algorithm also actually analyzes the frequency range and only allows you to determine (with a certain degree of probability) whether MPEG compression was applied at any of the encoding stages. The algorithm is tailored for MP3, so it is easy to “deceive” it with the help of Vorbis, AAC and Musepack codecs, so even if the program writes “100% CDDA”, this does not mean that the encoded audio is 100% identical to the original one.

And returning directly to the spectra. There is also a popular desire among some “enthusiasts” to disable the lowpass filter in the LAME encoder at all costs. Here there is a clear lack of understanding of the principles of coding and psychoacoustics. First, the encoder trims high frequencies with only one purpose - to save data and use it to encode the most audible frequency range. Extended frequency range can have a fatal impact on overall sound quality and lead to audible encoding artifacts. Moreover, turning off the cutoff at 20 kHz is generally completely unjustified, since a person simply cannot hear frequencies higher.

There is a certain “magic” equalizer preset that can significantly improve the sound

This is not entirely true, firstly, because each individual configuration (headphones, acoustics, sound card) has its own parameters (in particular, its amplitude-frequency characteristic). And therefore, each configuration must have its own, unique approach. Simply put, such an equalizer preset exists, but it differs for different configurations. Its essence lies in adjusting the frequency response of the path, namely, in “leveling out” unwanted dips and surges.

Also among people far from direct work With sound, setting a graphic equalizer with a “tick” is very popular, which actually represents an increase in the level of low-frequency and high-frequency components, but at the same time leads to muffling of vocals and instruments, the sound spectrum of which is in the mid-frequency region.

Before converting music to another format, you should decompress it to WAV

Let me immediately note that WAV means PCM data (pulse code modulation) in the WAVE container (file with *.wav extension). This data is nothing more than a sequence of bits (zeros and ones) in groups of 16, 24 or 32 (depending on the bit depth), each of which represents binary code the amplitude of the corresponding sample (for example, for 16 bits in decimal notation these are values from -32768 to +32768).

So, the fact is that any sound processor - be it a filter or an encoder - usually works only with these values, that is only with uncompressed data. This means that to convert audio from, say, FLAC to APE, you simply necessary First decode FLAC to PCM, and then encode PCM to APE. It's like repacking files from ZIP to RAR, you must first unpack the ZIP.

However, if you use a converter or just an advanced console encoder, the intermediate conversion to PCM occurs on the fly, sometimes without even writing to a temporary WAV file. This is what misleads people: it seems that the formats are converted directly from one to another, but in fact such a program must have an input format decoder that performs an intermediate conversion to PCM.

Thus, manual conversion in WAV will give you absolutely nothing but a waste of time.

There's a lot of talk these days about how we've lost real music with the advent of compressed audio formats like MP3, AAC and the like. Is this really true? Will Lossless formats save music? Can an untrained listener even distinguish music in MP3 from FLAC formats? Let's look into this issue.

What is bitrate?

You've probably heard the term "bitrate" before, and you probably have general idea about what it means, but it might be a good idea to look at the official definition so you know how it all works.

Bitrate is the number of bits or amount of data that is processed during certain period time. In audio, this usually means kilobits per second. For example, the music you buy from iTunes is 256 kilobits per second, meaning every second of the song contains 256 kilobytes of data.

The higher the bitrate of a track, the more space it will take up on your computer.. Typically, an audio CD takes up quite a lot of space, so it has become common practice to compress these files so that you can burn more music onto your HDD(or iPod, Dropbox or whatever). This is where the “lossless” and “lossy” formats come into debate.

Lossless and Lossy formats: what is the difference?

When we say "lossless" we mean that we haven't really changed original file . That is, we copied the track from the CD to our hard drive, but didn't compress it to the point that we lost any data. This is essentially the same as original track CD.

However, more often than not you probably rip your music into Lossy format. That is, you took a CD, copied it to your hard drive, and compressed the tracks so they didn't take up much space. A typical album is probably 100MB or so. The same album in a lossless format such as (also known as Apple Lossless) will take up about 300 MB, so it has become common practice to use lossy formats for more fast loading and greater hard drive savings.

The problem is that when you compress a file to save space, you remove chunks of data. Just like when you take an image from high quality, and compress it into a JPEG, your computer takes the original data and "tricks" certain parts of the image into making it look basically the same, but with some loss of clarity and quality.

Let's take the two images below as an example: The one on the right is clearly compressed and the quality has been reduced as a result.

Remember that you save space on your hard drive by compressing music in Lossy formats, which may have great importance for an iPhone with 32 GB of memory, but in terms of volume/quality ratio this is just a compromise.

There are different levels of compression: 128Kbps, for example, takes up very little space, but will also have lower playback quality than a larger 320Kbps file, which in turn is lower quality than the 1.411Kbps reference file. 1.411 Kbps is Audio CD quality, which is more than enough in most cases.

The whole problem is not how much the music is compressed, but what equipment you listen to it on.

Does bitrate really matter?

As memory becomes cheaper every year, listening to audio at higher bitrates, or even in Lossless formats, is starting to become more and more popular. But is it worth the time, effort and storage space on your phone or computer?

I don't like to answer questions this way, but unfortunately the answer is: it depends.

Part of the equation is the equipment you use.. If you use a quality pair of headphones or speakers, you are used to high frequency and dynamic range. So, you will most likely notice the disadvantages that come with compressing music into lower bitrate files. You may notice that low-quality MP3 files lack a certain level of detail; Thin background tracks may be more difficult to perceive, top and low frequencies won't be as dynamic, or you may hear distortion in the lead singer's vocals. In these cases, you may need a higher bitrate track.

However, if you listen to your music using a pair of cheap headphones on your iPod, you probably won't notice the difference between a 128Kbps file and a 320Kbps file, let alone 1.411Kbps lossless music. Remember when I showed you the image a few paragraphs above and noted that you probably had to look closely at it to see the flaws? Your headphones are like a truncated version of the image: they will make these imperfections difficult to perceive because they are physically unable to play the music for you the way you want.

The other part of the equation is, of course, your own ears. Some people may find it very difficult to distinguish between two different bitrates for the simple reason that they don't listen to much music. Hearing skills, like any other, develop with practice. If you listen to your favorite music often and a lot, your hearing becomes more accurate and begins to pick up small details and midtones. But until then, doesn't it really matter what bitrate you use?

So which format and bitrate should you choose for yourself? Is 320 Kbps enough for you, or do you definitely need the Lossless format?

The thing is, it's hard to hear the difference between a lossless file and a 320Kbps MP3 file. To hear the difference, you'll need some serious, high-quality equipment, good hearing, and a certain type of music (such as classical or jazz).

For the vast majority of people, 320 Kbps is more than enough for listening.

What else needs to be considered?

Recorded music can be helpful. Lossless files are more future-proof in the sense that you can always compress them to Lossy format when you need it, but you can't do the opposite and restore original CD quality from an MP3 file. This, again, is one of fundamental problems with online music stores: if you've built up a huge library of music in iTunes and one day decide you need more bitrate, you'll have to buy it again, only this time in CD format.

Whenever possible, I always buy or copy music in Lossless format for backup purposes.

I understand that for audiophiles, this is like a needle under your nails. As I said before, it all depends on you, your hearing and the equipment you have.

Compare two tracks recorded in Lossless and Lossy formats. Try a few different audio formats, listen to them for a while and see if it makes a difference for you or not.

In the worst case, you will spend several hours listening to your favorite music - not so bad, right? Enjoy it!