Hardware Acceleration for M4A Encoding and Decoding

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Hardware Acceleration for M4A Encoding and Decoding

Let’s talk about hardware acceleration for M4A encoding and decoding. Hardware acceleration uses specialized hardware to speed up M4A audio encoding and decoding, which is essential for fast audio processing. As a specialist in audio encoding, I’ve seen firsthand how much of an impact this can have on audio workflows. When your computer uses the specialized hardware to do these tasks instead of doing all of the work on the main processor, it is much more efficient, which results in faster processing and less power usage. I’ll explain how hardware acceleration works and why it’s very beneficial for M4A audio, using simple and easy-to-understand examples.

Understanding Hardware Acceleration

Hardware acceleration is like having a specialized tool for a specific job, and I’ve seen how it can make a huge difference in speed compared to using the general tools. Instead of using the main processor of the computer (the CPU) for all tasks, specialized hardware (like a GPU or a dedicated audio chip) does the processing. This can greatly reduce the workload on the CPU, making the whole process much faster. It’s like having a group of experts working together to do the job much faster, instead of relying on just one person to do it all. This is very helpful for audio encoding and decoding because they involve a lot of calculations.

Dedicated Hardware

Hardware acceleration uses dedicated hardware like GPUs or specific audio chips, designed to perform specific tasks very efficiently.
It’s like having a specialized car for racing; it goes much faster because it is designed for speed.

Reduced CPU Load

Hardware acceleration reduces the load on the CPU, so your computer can do other tasks smoothly while the audio is being encoded or decoded.
This is like having a helper who does the heavy work so you can do other things at the same time.

Increased Processing Speed

Hardware acceleration results in much faster encoding and decoding speeds compared to using software-based methods.
This can speed up your work, since the audio files are processed much faster thanks to the specialized hardware.

The Role of the CPU in M4A Processing

The CPU, or Central Processing Unit, is the main brain of your computer, and I view it as the most versatile, but not always the most efficient processor. When encoding or decoding M4A files using software methods, the CPU does all the calculations, and this can take a lot of its power. While CPUs can handle all tasks, they are usually not the fastest option for very demanding tasks, such as audio encoding and decoding, since it needs to do all of the work by itself. The CPU is a generalist that does everything but not always with the best performance.

General-Purpose Processing

CPUs are designed to handle a wide variety of tasks, from simple calculations to complex software applications, but they are not designed to do one thing really fast.
It is like having a general-purpose tool that can do many things, but it’s not the best tool for each of them.

Software-Based Encoding

When encoding and decoding audio in software, all the work is done on the CPU. This can be slow for complex operations.
Software-based encoding is very versatile, but may be very slow and power hungry compared to hardware alternatives.

Resource Bottleneck

When a CPU does all the encoding or decoding, it can become a bottleneck that slows down your computer.
The CPU has limited processing power and cannot always keep up with very demanding tasks, like audio processing.

GPUs and M4A Encoding

GPUs, or Graphics Processing Units, are designed for parallel processing, and I have seen that they are extremely efficient at tasks like audio encoding, and decoding. While they are mainly designed for graphics, GPUs can also be used for audio processing due to their ability to perform many calculations at the same time. This is very helpful for M4A encoding, since it involves a lot of similar calculations that can be done at the same time. Using GPUs for M4A encoding and decoding can greatly speed up the process.

Parallel Processing

GPUs can perform multiple calculations at the same time, which makes them very efficient for tasks like audio processing that require a lot of calculations.
It’s like having many workers doing different parts of the job at the same time, which results in much faster processing.

Offloading from CPU

Using the GPU for audio encoding or decoding frees up the CPU to perform other tasks, which makes the computer much more responsive.
This is like delegating tasks to other people, which results in less workload for you, and lets you work on other things.

Faster Encoding Times

GPUs can encode and decode audio much faster than CPUs, because they are designed to perform many similar calculations at the same time.
The speed improvements are very significant, and they can greatly reduce the encoding times.

Dedicated Audio Chips

Dedicated audio chips are specifically designed for audio processing, and I have seen how they can provide the very best results for audio tasks. These chips are optimized to encode and decode audio, with a very low latency, and very high efficiency. This means that these chips are the most efficient hardware option for audio processing. These chips can improve both speed and quality, making them the best option when these two are a concern.

Specialized for Audio

Dedicated audio chips are designed specifically for audio tasks, and they offer much better performance than a general-purpose processor.
These chips are optimized to do audio processing much faster and more accurately.

Low Latency Performance

These chips provide a low latency which is important for real time audio processing.
Low latency means less delays in processing the audio, which is important for audio tasks.

High Efficiency

Dedicated audio chips are designed to be very efficient, with low power consumption, and faster audio processing.
This makes them a good option for both portable and stationary devices, where efficiency is important.

Hardware Acceleration Benefits for M4A

Hardware acceleration provides several key benefits for M4A encoding and decoding, and from my work in the audio world I’ve seen these benefits in real world situations. These advantages include faster processing, better efficiency, and reduced power consumption. These benefits make hardware acceleration a great choice for all types of M4A audio projects. Hardware acceleration improves the overall performance, both for professional and home users.

Reduced Encoding/Decoding Times

Hardware acceleration significantly reduces the time to encode and decode M4A files, which allows users to process large audio files much faster.
This speeds up the audio workflows, which is very important when time is important.

Improved Efficiency

Hardware acceleration is more efficient than software based processing, and allows the CPU to focus on other tasks.
Hardware acceleration allows for more efficient processing, with less impact on the CPU.

Lower Power Consumption

Using specialized hardware consumes less power than software processing, this is very useful for portable devices where battery life is a concern.
Hardware acceleration is a great option to save energy and improve battery life.

How Hardware Acceleration Works in M4A

Hardware acceleration works by offloading some of the processing tasks to dedicated hardware components, and I’ve always been amazed by how this approach improves the audio performance. Instead of relying solely on the CPU, the software will use specialized units such as GPUs or dedicated audio chips, to do the audio processing tasks. This offloading process improves speed, and it reduces the burden on the main processor, making it work much faster and more efficiently. This allows the computer to work better and faster, and also saves power.

Offloading Processing

Hardware acceleration offloads the most demanding processing tasks to specific hardware, leaving the CPU free for other operations.
This method distributes the work across different specialized processing units, which improves speed and efficiency.

Direct Access to Hardware

Software can directly access the specialized hardware to perform encoding and decoding operations.
This avoids the overhead of the software processing which can be very slow and demanding.

Optimized Data Flow

Hardware acceleration provides an optimized data flow between the different components, making the overall process much more efficient.
This efficient data flow will result in a very fast and efficient encoding and decoding process.

Real-World Applications

Hardware acceleration is very useful in many real-world applications that require very fast audio processing. I’ve seen its power in various projects. For example, live audio processing benefits greatly from the reduced latency provided by hardware acceleration. When editing large audio files, the encoding and decoding process is much faster, and the time to save the files is greatly reduced. The benefits of hardware acceleration are useful in all audio situations where speed is important.

Live Audio Processing

Live audio processing requires very low latency and high processing speeds, and hardware acceleration makes this possible.
Hardware acceleration allows for real time audio processing with minimal delay.

Audio Editing

When working with large audio files, hardware acceleration speeds up the encoding and decoding process, which improves the overall workflow.
Thanks to hardware acceleration, the audio editing process is much more fluid.

Mobile Audio Devices

Mobile audio devices benefit greatly from hardware acceleration because of its low power consumption and high efficiency.
Battery life can be greatly improved with the use of hardware acceleration in portable devices.

Choosing Hardware for M4A Acceleration

Choosing the right hardware for M4A acceleration depends on specific needs and resources. In my opinion, there is not a single perfect solution, and the best hardware depends on the specific task and the required speed and quality. If speed is paramount, a good GPU may be the best choice. If the main concern is for real time audio, dedicated audio chips will be more suitable. Understanding the available options can help to make the best decision.

GPUs for M4A Processing

GPUs are a good choice for their parallel processing capabilities which are very helpful in speeding up M4A encoding and decoding.
GPUs can greatly improve processing speed, but they consume more power than other options.

Dedicated Audio Chips

Dedicated audio chips provide excellent performance with low latency and high efficiency, and are best for low latency applications.
They are a great option when the main concern is a low latency performance for audio processing tasks.

Integrated Hardware

Many modern devices include integrated hardware for audio processing, and these can also be a good option for those who don’t need extreme performance.
Integrated hardware offers a good balance between performance, power consumption and cost.

Latest words on Hardware Acceleration for M4A Encoding and Decoding

Hardware acceleration is essential for modern audio processing, particularly for M4A encoding and decoding. From my experience, it greatly enhances processing speed, efficiency, and power consumption. Using GPUs or dedicated audio chips can significantly improve the overall workflow. Tools like Mp4Gain can help you with your audio needs. Hardware acceleration is vital in our daily audio processing work, and I am sure that this technology will continue to evolve. Now, you have a good understanding of what hardware acceleration is and how it can greatly improve your audio experience.

What is hardware acceleration in audio processing?

Hardware acceleration uses specialized hardware, such as GPUs or dedicated audio chips, to speed up tasks like audio encoding and decoding. This allows to offload the work from the main CPU, making the computer work much faster and with better efficiency.

How does the CPU handle M4A encoding and decoding?

The CPU handles M4A encoding and decoding through software-based methods, performing all the calculations with its general-purpose architecture. While CPUs can do all of these tasks, they are not optimized for very demanding tasks, and can be very slow for complex audio encoding.

How do GPUs speed up M4A encoding and decoding?

GPUs speed up M4A encoding and decoding through their parallel processing capabilities, where they perform multiple calculations simultaneously. GPUs are very efficient doing this, which results in much faster processing than CPUs, and also a much more efficient workflow.

What are dedicated audio chips and how do they benefit audio tasks?

Dedicated audio chips are specifically designed for audio processing, and they provide low latency, high efficiency, and very fast audio encoding and decoding. These chips offer a much better performance than general purpose processors, like a CPU, which makes them ideal for audio processing tasks.

What are the key benefits of using hardware acceleration for M4A files?

The main benefits of hardware acceleration include faster encoding and decoding times, better processing efficiency, and lower power consumption. This helps to speed up the audio workflow, making all the audio tasks much faster. Using specialized hardware is very useful for large projects, since it saves a lot of processing time.

How does hardware acceleration offload tasks from the CPU?

Hardware acceleration offloads audio processing tasks to specialized components like GPUs or dedicated audio chips. This reduces the workload on the CPU, which then focuses on other tasks. This allows the CPU to work more efficiently, and perform other operations at the same time.

How does direct hardware access improve audio processing?

Direct hardware access allows software to use specialized hardware directly for encoding and decoding, which avoids the overhead of software processing. This process is much faster, and the software can access the full power of the specialized hardware. Direct hardware access results in faster processing times and better performance.

Why is low latency important for live audio processing?

Low latency means less delay in processing, which is essential for live audio processing applications, since any delay will be very noticeable by the users. Real-time audio requires very fast processing without any delays, and this is achieved with the right hardware and low latency performance.

How does hardware acceleration benefit mobile audio devices?

Hardware acceleration is very beneficial for mobile devices because it offers low power consumption, high efficiency, and faster processing times. This is very useful for portable devices where battery life is very important. Hardware acceleration can help extend battery life and improve the user experience in portable devices.

What is the best hardware option for M4A encoding and decoding?

The best hardware option depends on specific needs, and if speed is the main priority, a good GPU may be the best option. If low latency is more important, dedicated audio chips are better. Integrated hardware offers a good balance between power, cost, and efficiency. It’s always about the specific needs of the project and the user. There is not a single best solution.

Comments:

This article explained everything about hardware acceleration in a very easy and simple way, I didn’t understand these things before, but now I know how to improve my audio processing workflow, thanks a lot!

-AudioNewbie

Great info, man, I always wondered how some programs encode audio so fast, but now I understand it is all about hardware acceleration. I will look for software that uses this, thanks!

-TechFan

This is a great article, but I would like a more detailed explanation of the low latency part, maybe some examples of different hardware and its latency. But very good explanation!

-LatencyLover

Awesome explanation of hardware acceleration, I work with audio and I learned a lot about all of this. Very good and detailed information, thanks for sharing it!

-AudioPro

Very easy to understand explanations, I am not a tech expert, and I understood everything perfectly. Great examples, I learned a lot! Keep up the good work!

-SimpleUser

This article helped me understand how my computer can encode audio so fast, and why some programs are faster than others. Thank you for all the information, it was very helpful!

-CodeStudent

This is a great site, always with the best and most informative articles. This information about hardware acceleration was awesome, I learned a lot! Thank you guys!

-KnowledgeSeeker

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MP3 finally goes into the public domain

MP3 finally goes into the public domain

mp3

Open Source

Mp3 Public Domain

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:

CBR 096 kbps: 14000 – 15000 Hz;
CBR 112 kbps: 15000-15600 Hz;
CBR 128 kbps: 16000 – 16500 Hz;
CBR 160 kbps: 16500-17500 Hz;
CBR 192 kbps: 18000-18700 Hz;
CBR 224 kbps: 19000-19400 Hz;
CBR 256 kbps: 19500-19700 Hz;
CBR 320 kbps: 20,000 – 21,000 Hz.

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.

Audio Coding: Secrets Revealed Part 2

Audio Coding: Secrets Revealed Part 2

Bit Depth

Bit depth

audio encoding

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 encoding: secrets revealed

Audio encoding: secrets revealed

Audio Encoding

Audio settings for video capture and transmission.

audio and video encoding

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.

Audio encoding and processing

Audio encoding and processing

Encoding

Sound information.

ENCODING

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):

16 bits × 24,000 × 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

Audio Coding: Secrets Revealed – Part 2

Audio Coding: Secrets Revealed – Part 2

AUDIO ENCODING

Audio settings for video capture and transmission.

AUDIO ENCODING

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 = 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

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.

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 encoding: secrets revealed

Audio encoding: secrets revealed

audio encoding

Audio settings for video capture and transmission.

AUDIO ENCODING

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.

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.

Digital audio encoding

Digital audio encoding

Digital audio encoding

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.

DIGITAL AUDIO ENCODING

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.