The Science of Audio EncodingThe Science of Audio Encoding
Audio encoding is the process of converting analog sound into digital data. This data can then be stored or transmitted in a variety of formats, such as WAV, MP3, or AAC.
There are two main types of audio encoding: lossless and lossy. Lossless encoding preserves all of the original sound data, resulting in high-quality audio but large file sizes. Lossy encoding removes some of the original sound data, resulting in smaller file sizes but lower sound quality.
The process of audio encoding can be divided into three main steps: sampling, quantization, and compression.
Sampling
The first step in audio encoding is sampling. In this step, the analog sound signal is converted into a series of discrete values. The number of times per second that the sound signal is sampled is called the sample rate. Higher sample rates result in more accurate representations of the original sound signal, but they also result in larger file sizes.
Quantization
The second step in audio encoding is quantization. In this step, each sample value is rounded to the nearest integer value. The number of bits used to represent each sample value is called the bit depth. Higher bit depths result in more accurate representations of the original sound signal, but they also result in larger file sizes.
Compression
The third and final step in audio encoding is compression. In this step, the digital audio data is compressed to reduce its file size. There are a number of different compression algorithms that can be used, each with its own advantages and disadvantages.
The most common compression algorithms for audio encoding are:
MP3: MP3 is a lossy compression algorithm that is widely used for storing and transferring audio files. MP3 files are typically much smaller than WAV files, while still providing good sound quality.
AAC: AAC is another lossy compression algorithm that offers better sound quality than MP3. AAC files are typically slightly larger than MP3 files, but they offer a noticeable improvement in sound quality.
FLAC: FLAC is a lossless compression algorithm that offers similar sound quality to WAV, but with much smaller file sizes. FLAC files are a good choice for people who want the best possible sound quality without sacrificing file size.
Final Words
Audio encoding is a complex process that involves converting analog sound into digital data. The quality of the audio that is encoded can be affected by a number of factors, including the sample rate, bit depth, and compression of the audio file.
If you are looking for the best possible sound quality, you should use a lossless audio format such as WAV or FLAC. However, if you need to store or transfer audio files over a network, you should use a lossy audio format such as MP3 or AAC.
Digital Audio Encoding is the process of converting an analog audio signal into a digital format, which can be stored, processed, and transmitted electronically. It involves the use of an Analog-to-Digital Converter (ADC) to sample and quantize the analog audio waveform into a series of binary numbers that can be interpreted by a digital device. The resulting digital audio data can then be compressed, processed, and transmitted over various digital platforms, such as the internet, CDs, DVDs, and other digital storage devices.
The Importance of Digital Audio Encoding
Digital Audio Encoding has revolutionized the way we consume and produce audio content. It has made it possible to store, edit, and transmit high-quality audio content with minimal loss of quality. Some of the benefits of digital audio encoding include:
Improved sound quality: Digital audio encoding allows for high-quality audio content that is free from the distortions and noise associated with analog audio.
Easy storage and transfer: Digital audio files can be easily stored and transferred over various digital platforms with minimal loss of quality.
Efficient compression: Digital audio files can be compressed into smaller file sizes without significant loss of quality, making it easier to store and transfer large audio files.
Greater accessibility: Digital audio content can be easily accessed over various digital platforms, including the internet, mobile devices, and other digital devices.
The Digital Audio Encoding Process
The Digital Audio Encoding process involves several steps, which include:
Sampling: The analog audio waveform is sampled at regular intervals using an Analog-to-Digital Converter (ADC).
Quantization: The sampled waveform is quantized, i.e., each sample is assigned a binary number that represents its amplitude value.
Encoding: The quantized samples are encoded into a digital format, such as WAV, MP3, or AAC.
Compression: The encoded digital audio file can be compressed using lossy or lossless compression algorithms to reduce its file size.
Lossy vs. Lossless Audio Compression
Lossy and lossless audio compression are two types of compression algorithms used in digital audio encoding. Lossy compression algorithms compress audio files by removing data that is deemed unnecessary or redundant. This results in a smaller file size but may result in a loss of audio quality. Lossless compression algorithms, on the other hand, compress audio files without any loss of quality. This results in a larger file size but maintains the original audio quality.
Bitrate and its Importance in Digital Audio Encoding
Bitrate is a measure of the amount of data used to represent each second of digital audio. It is measured in bits per second (bps) or kilobits per second (kbps). The bitrate of a digital audio file has a significant impact on its quality and file size. Higher bitrates result in higher quality audio files but also larger file sizes. Lower bitrates result in smaller file sizes but may result in a loss of audio quality.
Common Digital Audio Formats
There are several digital audio formats used in digital audio encoding, including:
WAV: WAV is a lossless audio format that is commonly used for storing high-quality audio content.
MP3: MP3 is a lossy audio format that is commonly used for compressing and storing digital audio files for playback on various digital devices.
AAC: AAC is a lossy audio format that is commonly used for compressing and streaming digital audio content over the internet.
FLAC: FLAC is a lossless audio format that is commonly used for storing high-quality audio content, similar to WAV.
Challenges in Digital Audio Encoding
Despite the many benefits of digital audio encoding, there are several challenges that must be addressed to ensure optimal audio quality. These challenges include:
Sampling rate limitations: The sampling rate of an ADC can affect the accuracy of the digital audio representation. Higher sampling rates generally result in higher accuracy, but also require larger file sizes.
Bit depth limitations: The bit depth of an ADC can affect the dynamic range and noise floor of the digital audio representation. Higher bit depths generally result in higher accuracy, but also require larger file sizes.
Compression artifacts: Lossy compression algorithms can introduce compression artifacts, such as distortion and noise, which can degrade audio quality.
Future Developments in Digital Audio Encoding
Digital Audio Encoding is an ever-evolving field, with ongoing developments aimed at improving audio quality, reducing file sizes, and enhancing accessibility. Some of the latest developments include:
High-resolution audio: High-resolution audio formats, such as MQA and DSD, offer even higher audio quality than standard digital audio formats.
Immersive audio: Immersive audio formats, such as Dolby Atmos and DTS:X, offer a more immersive listening experience by incorporating height and surround sound elements.
Object-based audio: Object-based audio formats, such as MPEG-H 3D Audio, offer greater flexibility in audio content creation and delivery by enabling individual audio objects to be separately mixed and streamed.
FAQs
1. What is digital audio encoding?
Digital audio encoding is the process of converting an analog audio signal into a digital format, which can be stored, processed, and transmitted electronically.
2. Why is digital audio encoding important?
Digital audio encoding has revolutionized the way we consume and produce audio content by providing improved sound quality, easy storage and transfer, efficient compression, and greater accessibility.
3. What are some common digital audio formats?
Some common digital audio formats include WAV, MP3, AAC, and FLAC.
4. What is the difference between lossy and lossless audio compression?
Lossy compression algorithms compress audio files by removing data that is deemed unnecessary or redundant, resulting in a smaller file size but may result in a loss of audio quality. Lossless compression algorithms compress audio files without any loss of quality, resulting in a larger file size but maintaining the original audio quality.
5. What is bitrate and why is it important in digital audio encoding?
Bitrate is a measure of the amount of data used to represent each second of digital audio. It is important in digital audio encoding because it has a significant impact on audio quality and file size.
6. What are some challenges in digital audio encoding?
Some challenges in digital audio encoding include sampling rate limitations, bit depth limitations, and compression artifacts.
7. What are some future developments in digital audio encoding?
Some future developments in digital audio encoding include high-resolution audio, immersive audio, and object-based audio.
8. What is the difference between a lossy and lossless audio format?
Lossy audio formats use compression algorithms to reduce file size, sacrificing some audio quality in the process. Lossless audio formats, on the other hand, use compression algorithms that do not compromise audio quality, resulting in larger file sizes.
9. What is a sampling rate and how does it affect audio quality?
A sampling rate is the number of times per second that an analog audio signal is measured and converted into a digital signal. The higher the sampling rate, the more accurately the digital signal represents the original analog signal, resulting in higher audio quality. However, higher sampling rates also require larger file sizes and more processing power.
10. What is bit depth and how does it affect audio quality?
Bit depth refers to the number of bits used to represent each audio sample in a digital audio file. A higher bit depth allows for a greater dynamic range and lower noise floor, resulting in higher audio quality. However, higher bit depths also require larger file sizes and more processing power.
11. What is lossless compression?
Lossless compression is a compression algorithm that reduces the size of a digital audio file without sacrificing any audio quality. This is achieved by identifying and removing redundant or unnecessary data in the audio file.
12. What is immersive audio and how does it enhance the listening experience?
Immersive audio is an audio format that uses spatial sound technology to create a more immersive listening experience. This is achieved by incorporating height and surround sound elements, which create a more three-dimensional soundstage. This allows for a more realistic and engaging listening experience, especially when combined with a surround sound system.
Conclusion
Digital audio encoding has revolutionized the way we produce and consume audio content, providing improved sound quality, easy storage and transfer, efficient compression, and greater accessibility. While there are some challenges to overcome, ongoing developments in high-resolution, immersive, and object-based audio formats promise to further enhance the digital audio experience.
References
Bosi, M., & Goldberg, R. (2012). Introduction to digital audio coding and standards. Springer Science & Business Media.
Thompson, J. (2013). Understanding digital audio. Focal Press.
Perhaps many did not think so, but the mp3 standard so well known to all had problems with the purity of patents. On April 23, 2017, the last patents expired and the format was finally free. Technicolor has officially stopped collecting royalties from manufacturers of software and embedded solutions.
License
Although hardware mp3 decoding is built into all other coffee machines, until recently its use in commercial projects required royalties from the developer: Fraunhofer Society. In 2005 alone, the amount paid was one hundred million euros. Most of the patents became invalid in the European Union in 2012. However, some of them continued to operate in the United States due to peculiarities of local law. What does this news bring to the community? At least now it will be possible to compile Gentoo and listen to music at the same time immediately on the base distribution. Many distributions will be able to provide support for the standard to the main repository. Now, for example, Ubuntu itself requires the installation of non-free components from a separate Ubuntu Restricted Extras meta-package to support mp3.
Bourbon vanilla vs vanillin
How does this standard, which has been the main standard in this area for 24 years, despite many more advanced free options? mp3 is in many ways similar in principle to its cousin in the photo world: JPEG. Due to the imperfection of our hearing aid and the peculiarities of psychoacoustics, it is possible to “discard” those parts of the audio spectrum that do not make a significant contribution to the musical pattern. In particular, in the illustration above, you can see how the amount of information encoded in the high-frequency region increases.
High frequencies are often sacrificed for the sake of preserving detail in the lower region – vocals, most instruments (thanks for the comment, KorDen32). Standard values of cutoff frequencies for the lame encoder:
The method can be compared to the creativity of flavor chemists. You’ve probably noticed that strawberry gum is very conventionally strawberry, and there isn’t enough lemon in synthetic lemon tea. Any natural flavoring composition contains dozens and even hundreds of chemical compounds. But the main core generally creates only a very limited amount. So, for example, vanillin defines most of the aroma of natural vanilla, and if you don’t appreciate the subtle nuances too much, the remaining components can be neglected. mp3 uses the same principles, removing insignificant portions of the spectrum. Most people cannot tell the lossless formats by ear from the normally encoded 320kbps mp3s, which saves a lot of space when storing your media library.
Along with the sample rate, there is the bit depth or depth of the sound. Bit depth is the number of bits of digital information to encode each sample. Simply put, the bit depth determines the “accuracy” of the input signal measurement. The larger the digit capacity, the smaller the error will be for each individual conversion from the magnitude of an electrical signal to a number and vice versa. With the smallest possible bit depth, there are only two options for measuring sound accuracy: 0 for full silence and 1 for full sound. If the bit width is 8 (16), then by measuring the input signal, 2 8 = 256 (2 16 = 65,536) different values can be obtained.
Bit depth is fixed in the PCM codec, but for codecs that assume compression (eg MP3 and AAC), this parameter is calculated during encoding and may vary from sample to sample.
Bitrate
Bit rate is an indicator of the amount of information that one second of sound encodes. The higher it is, the less distortion and the closer the encoded composition is to the original. For linear PCM, the bit rate is very easy to calculate.
bitrate = sample rate × bit depth × channels
For systems like the Epiphan Pearl Mini that encode 16-bit (16-bit) linear PCM, this calculation can be used to determine how much additional bandwidth the PCM audio might require. For example, for stereo (two channels), the signal is digitized at 44.1 kHz at 16 bits and the bit rate is calculated as follows:
44.1 kHz × 16 bit × 2 = 1411.2 kbps
Meanwhile, audio compression algorithms like AAC and MP3 have fewer bits to transmit the signal (that’s their purpose), so they use low bit rates. Typically, the values are in the range of 96 kbps to 320 kbps. For these codecs, the higher the bit rate you choose, the more audio bits you get per sample and the better the sound quality.
Sample rate, bit depth and bit rates in real life.
Audio CDs, one of the most popular early inventions for the general public for storing digital audio, used 44.1 kHz (20 Hz – 20 kHz, human ear range) and 16 bits. These values were chosen to be able to save as much audio as possible to disk with good sound quality.
When video was added to audio and DVD and then Blu-ray discs came along, a new standard was created. DVD and Blu-Ray recordings typically use 48 kHz (stereo) or 96 kHz (5.1 surround) linear PCM format and 24-bit depth. These settings have been selected as ideal for keeping audio in sync with video while obtaining the best possible quality using the additional available disk space.
Our recommendations
CDs, DVDs, and Blu-Ray discs all have one goal: to provide the consumer with a high-quality playback engine. The goal of all developments was to provide high-quality audio and video without worrying about file size (if only it could fit on disk). Such quality could be provided by linear PCM.
In contrast, mobile media and streaming media have a completely different goal: to use the lowest bit rate, as low as possible, while still being sufficient to maintain acceptable quality for the listener. Compression algorithms are best suited for this task. You can follow the same principles for your records.
When recording audio from a video …
In case the record is used for the next on-ra-ki-bot, choose the 48 kHz PCM codec and the maximum bit depth (16 or 24) to provide the best audio quality. We recommend these parameters for Epiphan Pearl Mini.
When streaming audio from video …
With streaming or recording for later translation, good sound can be obtained with less bandwidth, using MP3 or AAC codecs with a frequency of 44.1 kHz and a bit rate of 128 kbit / s or higher. These parameters ensure that the sound is good enough without affecting the quality of the transmission.
Audio settings for video capture and transmission.
As people directly related to the AV sphere, we constantly talk about audio coding and audio codecs, but what is it? An audio codec is essentially a device or algorithm that can encode and decode a digital audio signal.
In practice, the audio waves that travel through the air are continuous analog signals. The signals are converted to digital form by a device called an analog-to-digital converter (ADC), and the reverse converter is called a digital-to-analog converter (DAC). The codec lies between these two functions and it is he who allows you to adjust some important parameters for the successful capture, recording and transmission of an audio signal: the codec algorithm, the sampling frequency, the bit width and the speed of the audio signal. data.
The three most popular audio codecs are Pulse-Code Modulation (PCM), MP3, and Advanced Audio Coding (AAC). The choice of codec determines the compression rate and the recording quality. PCM is a codec used by computers, CDs, digital phones, and sometimes SACD. The PCM signal source is sampled at regular intervals, and each sample is the digital amplitude of the analog signal. PCM is the simplest option for digitizing an analog signal.
With the correct parameters, this digitized signal can be completely converted back to analog without any loss. But this codec, which provides an almost complete identity with the original audio, is unfortunately not very cheap, which translates into very large file sizes, and such files are not suitable for streaming. We recommend using PCM to record digital images for your sources or when doing audio post-processing.
Fortunately, we always have the option of choosing a different codec that can compress digital data (rather than PCM) based on some helpful observations on the behavior of sound waves. But in this case, you have to make a compromise: all alternative algorithms are associated with “losses”, since it is impossible to completely restore the original signal, but nevertheless the result is still so good that most users will not be able to to catch the difference.
MP3 is an audio encoding format that uses a digital data compression algorithm that allows you to save the audio signal in smaller files. The MP3 codec is the most used by users to record and store music files. We recommend using MP3 to stream audio content as it requires less network bandwidth.
AAC is a newer audio encoding algorithm that is the successor to MP3. AAC has become the standard for MPEG-2 and MPEG-4 formats. In fact, this is also a digital data compression codec, but with less quality loss than MP3 when encoded with the same bit rate. We recommend using this codec for online streaming.
Sampling frequency (kHz, kHz)
Sample rate (or sample rate): the frequency with which the signal is digitized, stored, processed, or converted from analog to digital. Time sampling means that the signal is represented by several of its samples (samples) taken at regular intervals.
Measured in hertz (Hz, Hz) or kilohertz (kHz, kHz,) 1 kHz equals 1000 Hz. For example, 44,100 samples per second can be labeled 44,100 Hz or 44.1 kHz. The selected sample rate will determine the maximum playback frequency and, as follows from Kotelnikov’s theorem, to fully restore the original signal, the sample rate must be twice the highest frequency in the signal spectrum.
As you know, the human ear is capable of picking up frequencies between 20 Hz and 20 kHz. Given these parameters and the values shown in the table below, you can understand why 44.1 kHz was chosen as the sampling frequency for CD and is still considered a very good frequency for recording.
There are several reasons for choosing a higher sample rate, although it may seem like a waste of time and effort to reproduce sound outside the range of human hearing. At the same time, 44.1 – 48 kHz will suffice for the average listener for a high-quality solution to most problems.
Sound is a wave that travels through air, water, or other medium with a continuously changing intensity and frequency.
A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of different volume and pitch. The higher the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound
The human ear perceives sound at a frequency of 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).
A person can perceive sound in a wide range of intensities, in which the maximum intensity is 10 14 times greater than the minimum (one hundred thousand billion times). A special unit “decibel” (dbl) is used to measure the volume of sound (Table 5.1). Decreasing or increasing the volume of the sound by 10 dB corresponds to a decrease or increase in the intensity of the sound by 10 times.
Table 5.1. Sound volume
Sonar Volume in decibels
Lower limit of human ear sensitivity 0
Whisper of Leaves 10
Conversation 60
Horn 90
Jet engine 120
Pain threshold 140
Sound time sampling. For a computer to process sound, a continuous audio signal must be converted to a discrete digital form using time sampling. A continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.
Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels.
Sampling frequency.
A microphone connected to the sound card is used to record analog sound and convert it to digital format. The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit time, that is, the sampling frequency. The more measurements that are made in 1 second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the dialogue signal.
The audio sample rate is the number of measurements of the volume of a sound in one second.
The audio sample rate can range from 8000 to 48000 sound volume measurements per second.
Audio encoding depth. Each “step” is assigned a specific value for the volume level of the sound. Loudness levels of sound can be viewed as a set of possible states N, for which a certain amount of information is needed to encode, which is called audio encoding depth.
Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.
If the known encoding depth, the number of digital audio volume levels can be calculated using the formula N = 2 I. Let the sound encoding depth be 16 bit, then the number of sound volume levels is:
N = 2 I = 2 16 = 65 536.
During the encoding process, each sound volume level is assigned its own 16-bit binary code, the smallest sound level will correspond to the code 0000000000000000 and the highest, 1111111111111111.
The quality of digitized sound. The higher the sound sampling frequency and depth, the better the digitized sound will sound. The lowest quality of digitized sound, corresponding to the quality of telephone communication, is obtained at a sampling rate of 8000 times per second, a sampling rate of 8 bits, and by recording an audio track (“mono” mode). The highest quality digitized audio, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits, and the recording of two audio tracks (“stereo” mode ).
It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file. It is possible to estimate the information volume of a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements in 1 second and multiplied by 2 (stereo sound):
Audio settings for video capture and transmission.
Sampling frequency (kHz, kHz)
Sample rate (or sample rate): the frequency with which the signal is digitized, stored, processed, or converted from analog to digital. Time sampling means that the signal is represented by several of its samples (samples) taken at regular intervals.
Measured in hertz (Hz, Hz) or kilohertz (kHz, kHz,) 1 kHz equals 1000 Hz. For example, 44,100 samples per second can be labeled 44,100 Hz or 44.1 kHz. The selected sample rate will determine the maximum playback frequency and, as follows from Kotelnikov’s theorem, to fully restore the original signal, the sample rate must be twice the highest frequency in the signal spectrum.
As you know, the human ear is capable of picking up frequencies between 20 Hz and 20 kHz. Given these parameters and the values shown in the table below, you can understand why 44.1 kHz was chosen as the sampling frequency for CD and is still considered a very good frequency for recording.
There are several reasons for choosing a higher sample rate, although it may seem like a waste of time and effort to reproduce sound outside the range of the human ear. At the same time, 44.1 – 48 kHz will suffice for the average listener for a high-quality solution to most problems.
Bit depth
Along with the sample rate, there is the bit depth or depth of sound. Bit depth is the number of bits of digital information to encode each sample. Simply put, the bit depth determines the “accuracy” of the input signal measurement. The larger the digit capacity, the smaller the error for each individual conversion from the magnitude of an electrical signal to a number and vice versa. With the smallest possible bit depth, there are only two options for measuring sound accuracy: 0 for full silence and 1 for full sound. If the bit width is 8 (16), then by measuring the input signal, 2 8 = 256 (2 16 = 65,536) different values can be obtained.
Bit depth is fixed in the PCM codec, but for codecs that assume compression (eg MP3 and AAC), this parameter is calculated during encoding and may vary from sample to sample.
Bitrate
Bit rate is an indicator of the amount of information that one second of sound encodes. The higher it is, the less distortion and the closer the encoded composition is to the original. For linear PCM, the bit rate is very easy to calculate.
bitrate = sample rate × bit depth × channels
For systems such as the Epiphan Pearl Mini that encode 16-bit (16-bit) linear PCM, this calculation can be used to determine how much additional bandwidth the PCM audio might require. For example, for stereo (two channels), the signal is digitized at 44.1 kHz at 16 bits and the bit rate is calculated as follows:
44.1 kHz × 16 bit × 2 = 1411.2 kbps
Meanwhile, audio compression algorithms like AAC and MP3 have fewer bits to transmit the signal (that’s their purpose), so they use low bit rates. Typically, the values are in the range of 96 kbps to 320 kbps. For these codecs, the higher the bit rate you choose, the more audio bits you get per sample and the better the sound quality.
Sample rate, bit depth and bit rates in real life.
Audio CDs, one of the most popular early inventions for the general public for storing digital audio, used 44.1 kHz (20 Hz – 20 kHz, human ear range) and 16 bits. These values were chosen to be able to save as much audio as possible to disk with good sound quality.
When video was added to audio and DVD and then Blu-ray discs came along, a new standard was created. DVD and Blu-Ray recordings typically use 48 kHz (stereo) or 96 kHz (5.1 surround) linear PCM format and 24-bit depth. These settings have been chosen as ideal for keeping the audio in sync with the video while obtaining the best possible quality using additional available disk space.
Our recommendations
CDs, DVDs, and Blu-Ray discs all have one goal: to provide the consumer with a high-quality playback engine. The goal of all developments was to provide high-quality audio and video without worrying about file size (if only it could fit on disk). Such quality could be provided by linear PCM.
By contrast, mobile media and streaming media have a completely different goal: to use the lowest bit rate possible, while still being sufficient to maintain acceptable quality for the listener.
Audio settings for video capture and transmission.
As people directly related to the AV sphere, we constantly talk about audio coding and audio codecs, but what is it? An audio codec is essentially a device or algorithm that can encode and decode a digital audio signal.
In practice, the audio waves that travel through the air are continuous analog signals. The signals are converted to digital form by a device called an analog-to-digital converter (ADC), and the reverse converter is called a digital-to-analog converter (DAC). The codec lies between these two functions and it is he who allows you to adjust some important parameters for the successful capture, recording and transmission of an audio signal: the codec algorithm, the sampling frequency, the bit width and the speed of the audio signal. data.
The three most popular audio codecs are Pulse-Code Modulation (PCM), MP3, and Advanced Audio Coding (AAC). The choice of codec determines the compression rate and the recording quality. PCM is a codec used by computers, CDs, digital phones, and sometimes SACD. The PCM signal source is sampled at regular intervals, and each sample is the digital amplitude of the analog signal. PCM is the simplest option for digitizing an analog signal.
With the correct parameters, this digitized signal can be completely converted back to analog without any loss. But this codec, which provides an almost complete identity with the original audio, is unfortunately not very cheap, which results in very large file sizes, and such files are not suitable for streaming. We recommend using PCM to record digital images for your sources or when doing audio post-processing.
Fortunately, we always have the option of choosing a different codec that can compress digital data (rather than PCM) based on some helpful observations on the behavior of sound waves. But in this case, you have to make a compromise: all alternative algorithms are associated with “losses”, since it is impossible to completely restore the original signal, but nevertheless the result is still so good that most users will not be able to to catch the difference.
MP3 is an audio encoding format that uses a digital data compression algorithm that allows you to save the audio signal in smaller files. The MP3 codec is the most used by users to record and store music files. We recommend using MP3 to stream audio content as it requires less network bandwidth.
AAC is a newer audio encoding algorithm that is the successor to MP3. AAC has become the standard for the MPEG-2 and MPEG-4 formats. In fact, this is also a digital data compression codec, but with less quality loss than MP3 when encoded with the same bit rate. We recommend using this codec for online streaming.
PC-based audio coding is based on the process of converting air vibrations into electrical current fluctuations and the subsequent sampling of an analog electrical signal.
The encoding and reproduction of audio information is carried out using special programs. The quality of reproduction of the encoded sound depends on the sampling frequency and its resolution (sound encoding depth – the number of levels).
Digital audio is an analog audio signal represented by discrete numerical values of its amplitude.
Sound digitization is a technology with a divided time step and subsequent recording of the values obtained in numerical form. Another name for digitizing audio is analog to digital audio conversion, which includes the following operations:
Bandwidth limiting is done by using a low pass filter to suppress spectral components that are more than half the sample rate.
Time sampling, that is, replacing a continuous analog signal with a sequence of its values at discrete moments of time: samples.
Level quantization is the replacement of the signal’s reference value with the closest value of a set of fixed values: quantization levels.
Encoding or digitization, as a result of which the value of each quantized sample is represented as a number corresponding to the ordinal number of the quantization level.
This is done as follows: a continuous analog signal is “cut” into sections with a sample rate, a discrete digital signal is obtained, which goes through the quantization process with a certain bit depth, and is then encoded, that is, it is replaced by a sequence of code symbols. To record sound in a 20-20,000 Hz frequency band, a sampling frequency of 44.1 and higher is required (today there are ADCs and DACs with a sampling frequency of 192 and even 384 kHz). To obtain a high-quality recording, 16-bit is sufficient, however, to expand the dynamic range and improve the quality of the sound recording, 24 (less often 32) bits are used.
Sound coding methods (of course an electrical signal coming from a microphone) are based on the fact that, theoretically, any complex sound can be decomposed into a sequence of simpler harmonic signals of different frequencies, each of which it is a sinusoid, called the spectrum of the original signal. The task of encoding sound, like any other analog signal, is to represent it in the form of another analog or digital signal, which is more convenient for its transmission or storage in each specific case. Real sound sources have a limited spectrum width, therefore, for encoding, transformation methods are used that transform the original signal into one, the spectrum of which is more suitable for transmission on the selected channel. Representing an analog signal as another analog signal is commonly referred to as modulation and digitally as encoding. This division is very arbitrary. An analog signal can be represented as a harmonic signal (that is, a sinusoid), the parameters of which change depending on the value of the original signal. In the event that the amplitude of the sinusoid changes with a change in the original signal, it is amplitude modulation (AM). If, depending on the value of the original signal, the frequency or phase of the sinusoid changes, we are dealing with frequency modulation (FM) or phase modulation (PM). Amplitude and frequency modulation, for example, is widely used to transmit sound by radio. These types of modulation, of course, are not the decomposition of the original signal into harmonics. The development of digital technology and the use of computer processing and information storage has led to the widespread use of pulse encoding or modulation methods. Such types of modulation are, for example, pulse code modulation, in which the value of the original signal at regular intervals is represented in code form. The vast majority of “computer sound” is precisely the recording of the binary code of the received signal in short equal time intervals, determined by the sampling frequency. For storage and transmission through communication channels, this signal is usually compressed (reducing the volume by discarding unnecessary or insignificant information). In addition to pulse code modulation, other types of digital modulation (pulse width, pulse frequency, etc.) are also used to encode sound.
Digital audio is an analog audio signal represented by discrete numerical values of its amplitude.
Sound digitization is a technology with a divided time step and subsequent recording of the values obtained in numerical form.
Another name for digitizing audio is analog to digital audio conversion.
Sound digitization involves two processes:
sample (sample) a signal over time
amplitude quantification process.
Meanwhile, there is no need to worry about it. ”
Discretization of time.
Meanwhile, there is no need to worry about it. ”
The time sampling process is the process of obtaining the values of the signal that is being converted, with a certain time step: the sampling step. The number of measurements of the magnitude of the signal, carried out in one second, is called the sampling frequency or the sampling rate, or sampling frequency (from the English “sampling” – “sampling”). The lower the sampling step, the higher the sampling frequency and the more accurate representation of the signal that we will obtain.
This is confirmed by Kotelnikov’s theorem (in foreign literature it is found as Shannon’s theorem, Shannon). According to him, an analog signal with a limited spectrum can be accurately described by a discrete sequence of values of its amplitude, if these values are taken with a frequency that is at least twice the highest frequency in the spectrum of the signal. That is, an analog signal in which the highest spectrum frequency is F m can be accurately represented by a sequence of discrete amplitude values if F d> 2F m is satisfied for the sampling frequency F d.
In practice, this means that for the digitized signal to contain information on the full audible frequency range of the original analog signal (0 – 20 kHz), it is necessary that the selected sample rate be at least 40 kHz. The number of amplitude measurements per second is called the sampling rate (if the sampling step is constant).
The main difficulty of digitization is the inability to record the measured signal values with perfect precision.
Analog to digital converters (ADC).
Meanwhile, there is no need to worry about it. ”
The above process of digitizing sound is done using analog-to-digital converters (ADCs).
This transformation includes the following operations:
Bandwidth limiting is done by a low pass filter to suppress spectral components that are more than half the sample rate.
Discretization in time, that is, substitution of a continuous analog signal with a sequence of its values at discrete moments in time: samples. This problem is solved by using a special circuit at the input of the ADC – a sample and hold device.
Level quantization is the replacement of the signal’s reference value with the closest value of a set of fixed values: quantization levels.
Encoding or digitization, as a result of which the value of each quantized sample is represented as a number corresponding to the ordinal number of the quantization level.
This is done as follows: a continuous analog signal is “cut” into sections with a sample rate, a discrete digital signal is obtained, which goes through a quantization process with a certain bit depth, and is then encoded, that is, it is replaced by a sequence of code symbols. To record sound in a frequency band of 20-20,000 Hz, a sampling frequency of 44.1 and higher is required (today there are ADCs and DACs with a sampling frequency of 192 and even 384 kHz). To obtain a high-quality recording, 16 bits are sufficient, however, to expand the dynamic range and improve the quality of sound recording, 24 (less often 32) bits are used.
Meanwhile, there is no need to worry about it. ”
Encoding methods.
Frequency modulation.
Sound coding methods (of course we mean the electrical signal coming from the microphone) are based on the fact that, in theory, any complex sound can be broken down into a sequence of the simplest harmonic signals of different frequencies, each one of which is a sinusoid, called the original signal spectrum. The task of encoding sound, like any other analog signal, is to represent it in the form of another analog or digital signal, more convenient for its transmission or storage in each specific case.
