DAT tapes have a sample rate of 48, What are Channels? Channels are the number of separate recording elements in the data. For a real quick example, one channel is mono and two channels are stereo. In this document, both single and dual channel recordings will be discussed. What is the data? The data is the individual samples. An individual sample is the bit size times the number of channels.
For example, a monaural single channel , eight bit recording has an individual sample size of 8 bits. A monaural sixteen-bit recording has an individual sample size of 16 bits.
A stereo sixteen-bit recording has an individual sample size of 32 bits. Samples are placed end-to-end to form the data. So, for example, if you have four samples s1, s2, s3, s4 then the data would look like: s1s2s3s4. What is the header?
The header is used to provide specifications on the file type, sample rate, sample size and bit size of the file, as well as its overall length. If you have additional information to correct description or other, contact us. Sometimes files with same extension may contains different extensions.
A reading software player, converter, editor, other parse file. As rule, file consists of data blocks. These blocks have identifiers. And the reading software recognize the block types. Sometimes the software check data integrity. If there are non-correct data, the software may to reject file opening depend on implementation. Size compressed file types are used for saving hard disk space.
Especially, it is actually for portable devices: digital audio players DAP , mobile phones, etc. Portable devices are able to playback multichannel files. But it is listened at stereo headphones, as rule. So multichannel records consume disk space to extra channels. The space extra size issue may be solving via downmixing audio files to stereo.
It is impossibly to get rid of jitter in real music systems. Because there are electromagnetic interference, non-stability of clock generators, power line interference issues. Quantization error cause non-linear distortions. It correlate with musical signal. Correlated distortions are considered as especially unwanted to perceived sound quality. Dither is extremely low level noise, that added to musical signal before ADC or before bit depth truncation prior to DAC.
To reduce noise in audible band, noise shaping may be applied. It looks like "pushing" of noise energy to upper part of frequency range. But the shaping demands of band reserve to the "pushing". Size compression of audio content is way to save space at hard disk or increase throughput in communication line. Compression is performed by encoder and decoder software. Lossless compression is size compression when input and output binary audio data content are identical.
Lossless formats have same sound quality. There is opinion, that different sound may be there. Some objective hypotheses exists too. But still no researches, that are famous to author. Lossless compression is size compression when input and output binary audio data content aren't identical. Different lossy formats look for minimal losses by psychoacoustic criteria. And these compression methods are based on various hypotheses.
As example, AAC format was developed to improve mp3 sound quality according newer knowledges about brain processing of sonic information [ 1 ]. From this point of view, mp3 and FLAC are "bitstream" too. As rule, higher stream volume for single codec give better sound quality. But, other hand, higher bitrate may lead to lesser channel number in fixed band width of digital interface. As example, stereo instead multichannel. AV users asks what is use PCM or bitstream to transmit data from player to audio-video receiver of home theater.
Otherwise, use bitstream codecs. Dolby is size compressed PCM. It used to transmit audio signal thru digital audio interfaces with lower speed. If compression is lossless, it is not matter Dolby or original PCM there. Lossy compressing cause some quality losses. Generally, it is impossible to say, the losses will audible or not. Because different hardware is used there. It is common PCM in audio. Sound quality mean distortion level. However, distortions may have different distribution by frequency and phase.
And distortions must be estimated in the light of psychoacoustics. Aliases distortion appear during analog-to-digital and digital-to-analog conversion. Sample rate define the alias period on frequency axis. The period is half of sampling rate. All audio content above the period should be removed to avoid of distortions of useful musical signal. The analog filter makes the removing. However, analog filter isn't steep. Bit depth define minimal noise level into record.
If recorded musical stuff will digitally processed gain increasing, equalization, level normalizing, other , noise floor of processed stuff should be below DAC noise level. In audio software, processing may be implemented in or bit float point formats. These formats have high precision low quantization noise and better overload abilities, than integer ones. As far as author know, DAC can't receive data in float point formats. These formats are rounded to integer into playback software to send to DAC.
DAC with sigma delta modulator are able to receive float point formats. But author know nothing about such real implementations. It give base to myth that Hz is maximally reasonable sample rate. And there is opinion, that higher sampling rates aimed for ultrasound playback, that we can't hear.
Nyquist theorem, indeed, says that analog sine may be coded to digital PCM and restored back to analog without loses. But it is ideal concept, that require infinite time of recording and playback and ideal brickwall filter. Narrow transient band is difficult for analog filter. Steeper digital filter, more intensive its ringing distortions.
Also may be technical resource limitations to build steep enough filter. Inside DAC upsampling with digital filter is used for proper filter work. But hardware may have calculation resource limitation to implement sophisticated filter. We know that human hear sonic in range To keep sound quality signal must be higher noise. We can take noise level about dB as allowable. Digital audio data may be corrupted in transmitting or at storage. It can be checked via checksum comparison.
Audiophile players are capable to bit-perfect playback of audio files: audio file content is sent to DAC without altering. The most common format is integer PCM. This is raw, uncompressed sample data where each sample is an integer. PCM stands for pulse code modulation. Similarly, PCM data can be defined using a floating point value for each sample, although this is technically considered a different format. Back in the late 80s Electronic Arts came up with a general container file format that could be used to store different types of data — audio, graphics, etc.
Microsoft then took this format, switched the byte order from big-endian to little-endian to better suit Intel processors, and dubbed it RIFF Resource Interchange File Format.
All multi-byte numbers in a RIFF file are stored as little-endian. Each chunk starts with an 8-byte header containing a 4-byte identifier code, and a 4-byte size field. This is followed by the chunk body. The identifier code, called a FourCC , is a sequence of 4 bytes. When each byte is interpreted as an 8-bit ASCII character, they typically form a human readable string. For example, 0x52 0x49 0x46 0x46 i. Since this is a raw sequence of bytes, the characters are case-sensitive. The size should not include the 8-byte header.
If a chunk body has an odd number of bytes, it must be followed by a padding byte with value 0. In other words, a chunk must always occupy an even number of bytes in the file. This is followed by child chunks, nested inside the parent RIFF chunk. At minimum, the child chunks must include a format chunk "fmt " and a data chunk "data" , and the format chunk must come before the data chunk. If the format code in the format chunk is not 1 see below , then there must also be a fact chunk "fact".
Other optional chunk types are also possible. Although in practice, it often is. In this case, it is "RIFF". Next is the size field, which is the size of the entire Wave file except for the 8-byte RIFF chunk header. The first 4 bytes following the header will identify the type of RIFF chunk. Immediately following that will be the child chunks. The format chunk describes the format that the samples in the data chunk are encoded in.
The exact structure of the format chunk depends on the value of the format code field. If the format code is 1 integer PCM , then the format chunk will only contain the fields above the dashed line in the diagram below. Good to Know The reason for the different types of extension is that the Wave format is a container for many different kinds of sample formats, and because the Wave format has evolved over time to support new formats.
Extra fields that are needed for one sample format might not be needed for another sample format. This also allows new fields to be added without having to change pre-existing Wave files.
While some of these fields have a large range of possible values, in practice there are only a few that will actually be used. For some background on what some of this terminology means, check out this blog post. Format Code — Indicates how the sample data for the wave file is stored. The most common format is integer PCM, which has a code of 1.
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