What audio formats are compatible with iPhone, iPad, iPod?


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What audio formats are compatible with iPhone, iPad, iPod?

Iphone

Now there are far fewer such questions on the net, but before in many forums people asked before buying an iPhone: “What audio formats are compatible with iPhone, iPad, iPod?”

Apple Music

iPhone and iPad support the following audio file formats:
AAC (8 to 320 kbps), AAC (from iTunes Store), HE-AAC, MP3 (8 to 320 kbps), MP3 VBR, Audible (formats 2, 3, 4, Audible Enhanced Audio, AAX and AAX +) , AIFF and WAV, Apple Lossless (ALAC).

Most of the time iPhone and iPad users prefer MP3 and ALAC (Apple Lossless) formats, which they download from trackers, so there is practically no problem to copy music to iPhone, iPad.

What is Apple Lossless (ALAC) and how is it different from FLAC?
A few separate words should be said about the rather unusual Apple Lossless (ALAC) – this is an analog of the FLAC audio codec. Apple Lossless was specially designed by Apple to ensure that the user can enjoy the highest quality music while keeping battery consumption within reasonable limits.

Apple Lossless (ALAC) does not require high performance, so you can listen to music without quality loss, even on old iPod Nano. Apple takes great care to ensure that its devices can work for a long time without recharging, which is why we have a FLAC analog in the person of Apple Lossless.


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In what format is it better to listen to music? PART 4

In what format is it better to listen to music? PART 4

Audio File Format

What has changed today

AUDIO FORMATS

A rare sound engineer makes a digital master recording (which is then played back on physical media), using modern technologies to the full. So the chance that a 24-bit track is actually only 16-bit is extremely high.

High-quality analog recording on high-end gear is even harder to find today, if only for fans of this sound. Such is, for example, Jack White, the former leader of the White Stripes. At the same time, some of his recordings reference lo-fi variations, and looking for the scandalous sonic characteristics of the song becomes something of a foodie treat.

If you imagine an ideal source, only the trained ear or listening on high-quality audio equipment will allow you to find a compressed file. And already based on this (and without forgetting perception), it is worth drawing the following conclusion:

AAC is necessary and sufficient for medium-priced equipment, in the absence of which (and in the absence of sources that can be encoded in AAC) – MP3 with a constant 320 kbps bit rate, created with the Lame 3.93 codec (recommended keys for decoding: -cbr -b320 -q0 -k -ms).

The exceptions are recordings originally recorded in high quality, say, recorded on DVD-Audio, SACD, or recordings originally collected in DSD (or similar format) with a high bit rate.

Although without losses it has some characteristics. And we will tell about them next time.

The author does not like Apple. The author greatly appreciates the achievements of the Fraunhofers and was greatly surprised to learn that AAC is his work. 🙂

In what format is it better to listen to music? Part 3

In what format is it better to listen to music? Part 3

audio formats

Due to its advanced age, MP3 has significant limitations: the bit depth can be 16-24 bits, the sample rate is expressed only in discrete values ​​(8, 11,025, 12, 16, 22.05, 24, 32, 44.1, 48), the bit rate is limited to 320 kbps. Also, in the normal version of MP3, the number of channels is limited to two.

audio formats

AAC
The same rake, only in profile. Also developed by the Fraunhofer Society. Later and uses a different, more modern psychoacoustic model. The publicly available information allows us to conclude: yes, they managed to improve their own creation.

Even with the simplest numbers, AAC is a more flexible format. The bit depth of the files obtained with the help of this development varies from 16 to 24, the sampling frequency, if desired, will also allow not to lose the sound image and is in the range of 8-192 kHz. The data stream is generally close to lossless formats (up to 512 kbps), while the maximum number of AAC file channels reaches 48.

Which format is definitely the best?
Considering that AAC is MP3 reinvented after a dozen years, then the choice is in its favor. If you want, it makes sense to only compare MP3 and OGG.

On the graphics – good AudioCD, compressed OGG with 350 kbps variable bit rate and MP3 using Lame. The lower the graph, the closer the sound is to the original. It turns out to be a very interesting image. Although MP3 has clearly cut the high frequencies, unlike OGG, in which you can see a blockage below 2 kHz.

The frequency-time distribution of sound does not speak of less interesting things. At a constant 320kbps bit rate, MP3 is almost identical to the original recording. Everything seems to fit now. But … In fact, everything is even more confusing.

Why use at a loss at all when there is no loss available?
Common sense.

The fact is that most analog recordings do not contain the amount of information that would need to be stored in high-quality formats. Don’t forget that the native sample rate for CD is 44.1 kHz, the quantization is only 16 bits.

The above graphics well demonstrate the high fidelity of MP3 streaming. But for an audio cassette, magnetic tape (unless of course it is a master tape), the characteristics of an audio CD are unattainable. And for mass studio equipment, the ability to record analog sound corresponding to AudioCD has appeared relatively recently. It makes no sense to digitize in FLAC (and even more so in WAV) a concert recording or a disc from the pre-digital era, especially those made with magnetic media. They do not contain those spectra and the amount of information that containers can store without compression.

In what format is it better to listen to music? Part 2

In what format is it better to listen to music? Part 2

Audio Formats

The reference value of the audible range for humans is 16 Hz to 20 kHz, but you cannot hear and be aware of all incoming sounds simultaneously.

audio files

Hearing is discreet and your hearing sensitivity is not linear.

Modern psychoacoustic models accurately assess human hearing and are constantly improving. In fact, despite the guarantees of music lovers, musicians and audiophiles, to the inexperienced middle ear, the initial appearance of MP3 in maximum quality has become extremely noticeable. There are exceptions, they cannot cease to exist. But they are not always easily noticed by blind listening.

Formats using psychoacoustic compression models
There are many of these formats for lossy audio compression. The most common today are the following.

OGG (Vorbis)
In general, a file with the * .ogg extension is a “container”: it can contain multiple sound recordings with their own tags and characteristics. Most of the time, the files stored in it are compressed with the Ogg Vorbis codec, although others can be used, including MP3 or FLAC.

Its main advantages include a wide range of possible parameters during encoding: the audio sampling frequency can reach 192 kHz, the bit depth is 32 bits. By default, OGG uses a variable bit rate (although this is not shown on the properties screen), which can go up to 1000 kbps.

MP3
Unlike the free OGG, MP3 was developed by the Fraunhofer Society, an association of German institutes for applied research, which is very important for modern acoustics. Among audiophiles, by the way, this is an extremely respected office, yet they don’t like to admit it. But its developments are closely watched.

Unlike OGG, it can have variable (VBR) and constant (CBR) bit rate. By the way, it was thanks to MP3 that it was discovered that not all recordings can be encoded with high quality with a variable bit rate (see the above reasons, the encoding algorithms and their results in this case may be different when encoding the same source ).

In what format is it better to listen to music?

In what format is it better to listen to music?

Lossy compression

Understanding digital audio formats is not easy. It is even more difficult to come to an unequivocal conclusion in which format it is better to listen to music.

Lossy Formats

If you look at the audio format comparison table on Wikipedia, your eyes will start to flutter with columns of silent numbers. Let’s try to find out what’s behind this.
In what format is it better to listen to music? Three lost whales
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Let’s make a reservation right away that the article talks ONLY about general characteristics and will not include some details. Moving forward, Lifehacker will conduct its own unbiased investigation. And today we will try to generalize the already known experience in one way or another.

There is an analog and a figure.

The analog is good, but short-lived and inconvenient. Therefore, analog media, despite high vinyl sales, will not be making a comeback.

Digital audio can be of three main types:

in a format that does not use compression;
in a format that uses lossless compression;
in a format that uses lossy compression.
At first glance, lossless formats are more promising. This is not always the case, as we will discuss in more detail in one of the following materials. Uncompressed formats make no sense other than storing the master recordings needed to create audio content. They are easier to restore. Storing and listening to home recordings is superfluous.

Of the many parameters of digital audio, the user must first be concerned with sample rate (the accuracy of digitizing an analog signal in time), bit depth (the accuracy of digitizing in amplitude – volume) , the bit rate (the amount of information contained in the file in terms of one second).

Today we will talk about lossy.

For compressed sound, the concept of the psychoacoustic model is very important – the ideas of scientists and engineers about how a person perceives sound. The ear perceives the entire spectrum of acoustic waves entering it. However, the brain processes the signals.

Lossy compression format at a glance

Lossy compression format at a glance

lossy compression

“As you know, the music we listen to consists of a set of signals, each of which has its own characteristics, including loudness.

LOSSY COMPRESSION

The human hearing aid is designed so that we do not distinguish or poorly distinguish a weak (low) signal from the background of a strong (strong) signal. This principle forms the basis of modern means of compression (compression) of audio data.

If we imagine that a signal of a certain length is divided into many parts, and each part is processed in such a way that a weaker signal, which is difficult to distinguish from the background of a strong one, falls under the knife, and one remains a signal louder, then this will be an approximate audio compression model. Consequently, the level of data compression will depend on how many parts (samples) the original file will be divided into and how many weak signals from each individual sample will be removed (what the bit rate will be, the number of bits in a sample). sample of a specified duration). This coding principle is called lossy coding or lossy coding.

Ogg Vorbis is a completely open and patent-free audio format that allows you to store and transmit audio information with high sound quality (44.1-48.0 kHz sample rate, more than 16 bits, polyphony (multi-channel audio) ) and bit rates ranging from 16 to 512 Kbps per channel. In this case, the number of processed channels can reach 255.

MP3 or MPEG-1 Layer 3 audio is by far the most popular format for storing and transmitting compressed data. This format was developed by the Fraunhofer Institut, Germany. “Http://ru.wikibooks.org/wiki/Compression_Audio_data_with_lossy

Comparative tests

Sound Forge 7.0 (Spectral Analysis / Spectrum Analysis function) was used for the analysis of the sound signal.

“Spectral analysis is a signal processing technique that can reveal the frequency content of a signal. Solving the problems of spectral analysis is possible through the use of the fast Fourier transform, which makes it possible to determine the contribution of individual components of the vibration spectrum to the overall vibration picture. “Http: //masters.donntu. edu.ua/2007/fema/belinskaya/library/a4/art4.htm

The following graphs were obtained in the form of an amplitude distribution in the frequency domain, the spectrum of the signal is presented using a Blackman-Harris / Blackman-Harris window and a maximum sampling frequency (FFT size) of 65536, this gives allows you to analyze the smallest details of the signal at frequencies around 20,000 Hz, without smoothing.

The analysis of the spectrum of the compressed signal assumes the presence of a recording of the original quality, for this we use a licensed audio CD made in the USA “Kevin Yost – Bongo Madness”, with standard characteristics 44100 Hz / 16 bit

The rich electronic sound spans the entire frequency spectrum and captures even the inaudible range (20,000 Hz to 22,000 Hz), as can be seen in the graph below. Considering that it is generally possible to notice codec compression at higher frequencies, the 10-20 kHz range will be considered.

History and characteristics of the MPEG standards. Part 5

History and characteristics of the MPEG standards. Part 5

mpeg

ABR: mechanism

Mpeg

Suppose user specified ABR mode and a certain bitrate B (user can specify absolutely any bitrate from 32 to 320, even not from standard bitrate grid, for example you can specify 129 as the rate Average Bit Rate). The encoder accepts a piece of audio (frame) to be encoded. In the same way, as in CBR, it determines its complexity (we will talk about this later). If the passage is complex, then the encoder also takes more bits for it, but not from the repository (as in CBR), but simply increasing the bitrate by the required number of steps (the selected bitrate must be included in the standard grid), thus creating a “virtual repository” (you can increase the bitrate here, this is not CBR). What does “virtual reservoir” mean? It’s simple: we assume that the user-specified bit rate B is not sufficient for the encoder, standard N bit rate, where: N> = K (we call this choice of bit rate “virtual deposit”). Then there is a K-bit encoding of the taken piece of audio. However, N> = K, that is, we use fewer bits than there are in the taken frame, so won’t we throw away these extra bits? It is these extra bits that we write to the actual deposit. Since ABR has the ability to use a “virtual reservoir”, it makes no sense to build a standard reservoir, so when the next piece of audio arrives, the bits from the reservoir will be used to encode it first, and then the encoder will decide what rate bit is needed next. In other words, if in CBR the encoder always tries to accumulate as many bits in the reservoir as possible, then in ABR the encoder, on the contrary, tries to get rid of the bits in the reservoir,

Simple passages are encoded with fewer bits, they take about 95% of the specified bit rate B, but now the rest is not deposited into the repository, the encoder just takes a frame with a lower bit rate. The resulting difference (the remaining bits) is written to the standard repository (don’t discard the remaining bits …). Example. Let’s say a “simple” passage has arrived. Then the encoder takes all the bits (if any) in the repository (present), then looks for the standard bitrate closest to which the total number of bits obtained for this frame (all the bits in the repository + rate of bits taken) is 95% of the user-specified bitrate B performs the encoding and the extra bits (if any) are stored back in the repository.

APR: Summary

So using a tank in ABR is different from CBR. In CBR, the bit rate cannot be changed, and the repository is specially saved by storing there the bits that were left (were saved) from the frame encoding at an initially fixed bit rate determined during a single pass; if bits are required for encoding and the repository is empty, then it is empty, nothing can be done about it, and encoding is simply done at the specified bit rate to the detriment of quality. In ABR, the bit rate is variable and the standard deposit is not really necessary, however, since the increase (decrease) of the bit rate necessarily occurs up to a certain table values ​​that can turn out to be higher than the number. of bits required by the encoder, then the extra bits, of course, are not discarded, but are stored in the repository. In other words, in CBR the accumulation of the standard pool is the main task, while in ABR there is an unlimited “virtual pool” and the standard is used only to store additional bits formed as a result of the difference between the table values. Bitrate and actually required bitrate.

Vbr

VBR: variable bit rate. The user indicates the desired quality. Lame, based on his psychoacoustic model, assigns to each frame exactly the number of bits necessary to achieve a certain quality. In the output stream, the frames have respectively different bit rates (which always fit into the standard bit rate table). Warehouse usage in VBR is absolutely identical to ABR, only unused frame queues go there.

Methods for estimating signal complexity

So the main difference between CBR, ABR and VBR, as you probably already understood from the above, is the use of different methods to calculate the number of bits needed to encode each frame.

History and characteristics of the MPEG standards. Part 4

History and characteristics of the MPEG standards. Part 4

MPEG Standards

What are the differences between CBR, VBR and ABR modes? (applied to the Lame encoder)

mpeg

Before starting the conversation, let’s clarify two details:

1. MP3 encoding occurs block by block: the encoded file is divided into frames (frames) with the same interval, each frame is encoded and written to the output stream; therefore, the output stream also has a frame structure.

2. Frames cannot be encoded at any bit rate, but only at one of the standard MPEG1 Layer III bit rates listed in the table: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320. The standard does not provide encoding at any intermediate bit rate (“free format”).

Introduction

People using VBR in Lame generally argue this with the phrase, “I want to get constant quality, not constant bitrate. In fact, in music there are simple passages, for which 128 Kbps is enough (for example, pauses between songs), and there are also complex passages, in which a person with good hearing, a good audio card and other audio equipment. audio will hear compression. defects even at 320 Kbps / sec. In fact, such an argument is not entirely valid.

CBR

Even in CBR mode, the mp3 encoder can reallocate bits over time, emphasizing more or fewer bits during complex or simple passages, thus improving the overall sound quality. This bit reassignment is done through the so-called bit deposit: during the encoding of simple passages, the encoder spends not the entire user-specified bit rate on them, but only about 90%, about 10% is Store in bin to code difficult spots (bin is empty initially). When encoding complex passages, the encoder will use all 100% of the specified bit rate and add extra bits from the bucket (if any, that is, if the bucket is not empty). Unfortunately, according to the standard, the size of the tank is limited. This means that if a single signal lasts long enough, the tank builds its volume up to certain maximum allowed limits, and then the encoding continues using all 100% bit rate. And the opposite situation: if a complex signal lasts long enough, all the saved bits are taken from the repository (gradually) and then encoding is done using now all 100% of the bit rate.

ABR: Explanation

One could say that the reservoir does a good job with its main function – accumulating “extra” bits during simple passages and issuing them as additional bits when encoding complex passages, if not for one “but”: it has a finite and, moreover, Very limited in size, which means that it can only be stored up to certain limits and consequently can also be removed until the tank is empty. It is to eliminate this major drawback of the tank that the ABR was developed.

The main difference between ABR and CBR is that in CBR all frames must be the same size (that is, the bit rate for all frames must be the same), but in ABR this limitation is removed, respectively, there is an opportunity to use an almost infinite tank instead of the standard, very limited in size. “virtual” reservoir. Does it look like this.

History and characteristics of the MPEG standards. Part 3

History and characteristics of the MPEG standards. Part 3

MPEG

3) The MPEG-4 standard is a special article. MPEG-4 is not just an algorithm for compressing, storing and transmitting video or audio information. MPEG-4 is a new way of presenting information, it is an object-oriented representation of multimedia data. The standard operates with objects, organizes hierarchies, classes, etc. from them, he builds scenes and controls their transfer.

MPEG

 

The objects can be ordinary audio or video streams, as well as synthesized audio and graphics data (voice, text, effects, sounds …). These scenes are described in a special language. We will not dwell on this standard in detail; this is a topic for a separate extensive discussion. It can only be said that as a means of audio compression in MPEG-4, a set of various audio coding standards is used: the MPEG-2 AAC algorithm, the TwinVQ algorithm, as well as HVXC (Excitation Coding) voice coding algorithms. harmonic vector) – for 2-4 Kbps bit rates and CELP (Code Excited Linear Predictive) – for 4-24 Kbps bit rates. In addition, MPEG-4 has many scalability mechanisms.

4) The MPEG-7 standard, the development of which has not yet been completed, is fundamentally different from all other MPEG standards. The standard is not being developed to establish a framework for transferring data or writing and describing data of any particular kind. The standard is intended to be descriptive, intended to regulate the characteristics of any type of data, even analog. The use of MPEG-7 is intended to be closely related to MPEG-4. MPEG-7 is scheduled for release in 2001.

For the convenience of handling compressed streams, all MPEG algorithms are designed in such a way that they allow decompression (retrieval) and playback of a stream simultaneously with its reception (download) – stream decompression “on the fly” (stream playback) . This opportunity is widely used on the Internet, where the speed of information transfer is limited, and with the use of these algorithms, it is possible to process the information at the moment it is received without waiting for the end of the transfer.

What are CBR and VBR?

As you know, the result of encoding a signal using an algorithm such as MPEG-1 Layer III (MP3) (or some other algorithms) is a bit stream with a frame (block) structure. This is due to the fact that the source stream is not encoded in its entirety, but in parts. That is, in fact, the original stream is divided into blocks of a certain fixed length, then each block (frame) is encoded individually, and the result (encoded information block) is sent to the resulting stream (either a file or a stream of data).

CBR (constant bit rate) is a method of encoding the original audio stream, in which all its blocks (frames) are encoded with the same parameters (with the same bit rate). In other words, the bitrate over the entire length (all frames) of the resulting stream is constant.

VBR (Variable Bit Rate) is a method of encoding the original audio stream, in which each separate block (frame) is encoded with its own bit rate. The choice of the optimal bit rate to encode a given frame is made by the encoder itself by analyzing the “signal complexity” in each individual frame.

History and characteristics of the MPEG standards. Part 2

History and characteristics of the MPEG standards. Part 2

MPEG Standards

2) The MPEG-2 standard was developed especially to encode TV signals from television broadcasts, therefore, we would not have stopped considering MPEG-2 if in April 1997 this set had not received a “continuation” in the form of MPEG- 2 AAC (MPEG-2 Advanced Audio Coding – Advanced Audio Coding) algorithm.

MPEG Video Standards - The Road From 1 to 21

 

The MPEG-2 AAC standard is a collaborative effort between the Fraunhofer Institute, Sony, NEC, and Dolby. MPEG-2 AAC is a receiver for MPEG-1 technology. There are several types of this algorithm: Homeboy AAC, AT&T a2b AAC, Liquifier AAC, Astrid / Quartex AAC, and Mayah AAC. The highest sound quality compared to MPEG-1 Layer III is provided by the two penultimate implementations. All previous versions of the AAC algorithm are not compatible with each other.

As with the standard MPEG-1 audio coding suite, the AAC algorithm is based on the analysis of psychoacoustic signals. At the same time, the AAC algorithm has many additions to its mechanism, aimed at improving the quality of the output audio signal. In particular, a different type of transformation is used, noise processing is improved, the filter bank is changed, and the way the output bit stream is recorded is improved. Furthermore, AAC allows you to store the so-called encoded audio signal in the encoded audio signal. “Watermarks”: copyright information. This information is embedded in the bit stream during encoding in such a way that it is impossible to destroy it without destroying the integrity of the audio data. This technology (under the Multimedia Protection Protocol) allows you to control the distribution of audio data (which, by the way, is an obstacle to the distribution of the algorithm itself and the files created with it). It should be noted that the AAC algorithm is not backward compatible (NBC – not backward compatible) with MPEG-1 levels, even though it is a continuation (refinement) of MPEG-1 Layer I, II, III.

MPEG-2 AAC provides three different encoding profiles: Main, LC (Low Complexity), and SSR (Scalable Sample Rate). Depending on the profile used during encoding, the encoding time and the quality of the resulting digital stream change. The main main profile provides the highest sound quality (at the slowest compression rate). This is due to the fact that the main profile includes all the mechanisms for analyzing and processing the input stream. The LC profile is simplified, which affects the sound quality of the resulting stream, greatly affects the compression rate, and more importantly, the decompression. The SSR profile is also a simplified version of the main profile.

Speaking of sound quality, we can say that the 96 Kbps AAC (main) transmission provides the same sound quality as the 128 Kbps MPEG-1 Layer III transmission. With 128 Kbps AAC compression, the sound quality is notably superior. to MPEG-1 Layer III 128 Kbps.