1. Digital Audio Processing
By: Eng. Mohanned Dawoud

2. Audio signal processing
Audio signal processing, sometimes referred to as audio processing.
It is the processing of a representation of auditory signals, or sound.

3. Analog signals An analog representation is usually electrical.
A voltage level represents the air pressure waveform of the sound.

4. Digital signals
A digital representation expresses the pressure wave-form as a sequence of symbols, usually binary numbers, which permits digital signal processing.
human ears cannot perceive frequencies below approx. 20 Hz or above approx. 18 kHz (strongly depends on the age of the listener). Therefore, there is no significant loss of information when the analog signal is sampled using a high enough sampling rate.

5. Digital signals
The signal-to-noise ratio (SNR), which measures the noise level, can be easily calculated thru this formula, where n is the number of bits used on the ADC:
SNR = 6.02 x n dB
The higher the SNR, the better. An 8-bit ADC provides a SNR of 49.9 dB, while a 16-bit SNR provides a SNR of 98 dB (which is, by the way, a virtually no-noise value).
More than 130 dB Signal-to-noise ratio is almost impossible to achieve.

7. Application areas Processing methods and application areas include:
Storage.
level compression
data compression.
Transmission.
Enhancement (e.g., equalization, filtering, noise cancellation, echo or reverb removal or addition, etc.)

8. Sound recording and reproduction
Sound recording and reproduction is the electrical or mechanical inscription and re-creation of sound waves, usually used for the voice or for music.

9. Sound recording and reproduction
The two main classes of sound recording technology are analog recording and digital recording.
Analog recording is achieved by a small microphone diaphragm that can detect changes in atmospheric pressure (acoustic sound waves) and record them as graphic sound waves on a medium.
Digital recording and reproduction uses the same analog technologies, with digitization of the sonographic data and signal.

10. Sound recording and reproduction
The most recent and revolutionary developments have been in digital recording, with the invention of purely electronic consumer recording formats such as the WAV digital music file and the compressed file type, the MP3.
use of computers has made editing operations faster and easier to execute with software, and the use of hard-drives for storage has made recording cheaper

11. Dynamic range compression
In music, dynamic range is the difference between the quietest and loudest volume of an instrument, part or piece of music.
Dynamic range compression, also called DRC (often seen in DVD player settings) or simply compression, is a process that reduces the dynamic range of an audio signal.
Compression is used during sound recording, live sound reinforcement, and broadcasting to control the level of audio. A compressor is the device used to apply compression.

12. Dynamic range compression
In simple terms, a compressor is an automatic volume control.
Loud sounds over a certain threshold are reduced in level while quiet sounds remain untreated (this is known as downward compression, while the less common upward compression involves making sounds below the threshold louder while the louder passages remain unchanged).
In this way it reduces the dynamic range of an audio signal.
This may be done for aesthetic reasons, to deal with technical limitations of audio equipment, or to improve audibility of audio in noisy environments.

13. Dynamic range compression
A compressor reduces the gain (level) of an audio signal if its amplitude exceeds a certain threshold.
The amount of gain reduction is determined by a ratio.
For example, with a ratio of 4:1, when the (time averaged) input level is 4 dB over the threshold, the output signal level will be 1 dB over the threshold. The gain (level) has been reduced by 3 dB. When the input level is 8 dB above the threshold, the output level will be 2 dB; a 6 dB gain reduction.
A more specific example for a 4:1 ratio:
Threshold = −10 dB
Input = −6 dB (4 dB above the threshold)
Output = −9 dB (1 dB above the threshold)

14. Dynamic range compression
Design
Feed-forward: is used today on all compressors.
The signal entering a compressor is split, with one copy sent to a variable-gain amplifier and the other to a path called the side-chain.
Control circuit calculates the required amount of gain reduction.
The control-circuit outputs the requested gain-reduction amount to the amplifier.
Feedback type: Early compressor designs were based on this type.
the signal feeding the control circuit was taken after the amplifier.

15. Dynamic range compression
Compressor features
Threshold
Threshold is the level above which the signal is reduced. It is commonly set in dB, where a lower threshold (e.g. -60 dB) means a larger portion of the signal will be treated (compared to a higher threshold of -5 dB).
Ratio
The ratio determines the input/output ratio for signals above the threshold.
The highest ratio of ∞:1 is commonly achieved using a ratio of 60:1.

16. Dynamic range compression
Attack and release
A compressor might provide a degree of control over how quickly it acts.
The 'attack phase' is the period when the compressor is increasing gain reduction to reach the level that is determined by the ratio.
The 'release phase' is the period when the compressor is decreasing gain reduction to the level determined by the ratio, or, to zero, once the level has fallen below the threshold.
The length of each period is determined by the rate of change and the required change gain reduction.

17. Dynamic range compression
Soft and hard knees
Another control a compressor might offer is hard/soft knee.
This controls whether the bend in the response curve is a sharp angle or has a rounded edge.
A soft knee slowly increases the compression ratio as the level increases and eventually reaches the compression ratio set by the user.
A soft knee reduces the audible change from uncompressed to compressed, especially for higher ratios where the changeover is more noticeable.

18. Dynamic range compression
Stereo Linking
Stereo linking can be achieved in two ways:
Either the compressor sums to mono the left and right channel at the input, then only the left channel controls are functional.
The compressor still calculates the required amount of gain reduction independently for each channel and then apply the highest amount of gain reduction to both .
Make up gain
Because the compressor is reducing the gain (or level) of the signal, the ability to add a fixed amount of make-up gain at the output is provided so that an optimum level can be used.

19. Dynamic range compression
Compressor usage
high threshold and low compression ratio achieve dynamic range reduction with few obvious effects so that the source material is being compressed very slightly most of the time.
To deliberately soften the attack of a snare drum, they might choose a fast attack time and a moderately fast release time combined with a higher threshold.
To accentuate the attack of the snare, they might choose a slower attack time to avoid affecting the initial transient. It is easier to successfully apply these controls if the user has a basic knowledge of musical instrument acoustics.
It should be noted that compression can also be used to lift the soft passages of a selection, pulling the sound toward a compressed "middle". Hence, loud sounds are pulled back and soft passages are boosted.

20. Dynamic range compression
Limiting
Compression and limiting are no different in process, just in degree and in the perceived effect.
A limiter is a compressor with a higher ratio, and generally a fast attack time.
Engineers sometimes refer to soft and hard limiting which are differences of degree. The "harder" a limiter, the higher its ratio and the faster its attack and release times.

21. Dynamic range compression
Side-chaining
Side-chaining uses the signal level of another input or an equalized version of the original input to control the compression level of the original signal.
This is used by disc jockeys to lower the music volume automatically when speaking.
A stereo compressor without a sidechain can be used as a mono compressor with a sidechain. The key or sidechain signal is sent to the first (main) input of the stereo compressor while the signal that is to be compressed is routed into and out of the second channel of the compressor.

22. Dynamic range compression
Parallel compression
parallel compression is to insert the compressor in a parallel signal path.
give a measure of dynamic control without significant audible side effects, if the ratio is relatively low and the compressor's sound is relatively neutral.
On the other hand, a high compression ratio with significant audible artifacts can be chosen in one of the two parallel signal paths.

23. Dynamic range compression
Multiband compression
Multiband compressors are compressors that can act differently on different frequency bands.
It is as if each bandpass has its own compressor with its own threshold, ratio, attack, and release.
Serial compression
Serial compression is a technique used in sound recording and mixing.
Serial compression is achieved by using two fairly different compressors in a signal chain. One compressor will generally stabilize the dynamic range while the other will more aggressively compress stronger peaks.

24. Dynamic range compression
Common uses
Public spaces
Music production
Voice
Broadcasting
Marketing

25. Audio compression (data)
Audio compression is a form of data compression designed to reduce the size of audio files.
Audio compression algorithms are implemented in computer software as audio codecs.
Generic data compression algorithms perform poorly with audio data.
Consequently, specific audio "lossless" and "lossy" algorithms have been created.
In both lossy and lossless compression, information redundancy is reduced. 

26. Audio compression (data)
Lossless audio compression
Lossless audio compression allows one to preserve an exact copy of one's audio files.
Compression ratios are similar to those for generic lossless data compression (around 50–60% of original size).
Compression ratios are substantially less for lossy compression (which typically yield 5–20% of original size).

27. Audio compression (data)
Use of Lossless audio compression
The primary use of lossless encoding are:
Archives : For archival purposes, one naturally wishes to maximize quality.
Audio quality: Being lossless, these formats completely avoid compression artifacts.

28. Audio compression (data)
Use of Lossless audio compression (cont.)
A specific application is to:
Store lossless copies of audio.
Produce lossily compressed versions for a digital audio player.
As formats and encoders improve, one can produce updated lossily compressed files from the lossless master.
As file storage and communications bandwidth have become less expensive and more available, lossless audio compression has become more popular.

29. Audio compression (data)
Formats of Lossless audio compression
Shorten was an early lossless format; newer ones include Free Lossless Audio Codec (FLAC), Apple's Apple Lossless, MPEG-4 ALS, Monkey's Audio, and TTA.
Some audio formats feature a combination of a lossy format and a lossless correction; this allows stripping the correction to easily obtain a lossy file. Such formats include MPEG-4 SLS (Scalable to Lossless), WavPack, and OptimFROG DualStream.
Some formats are associated with a technology, such as:
Direct Stream Transfer, used in Super Audio CD
Meridian Lossless Packing, used in DVD-Audio and Dolby TrueHD, used in in Blu-ray and HD DVD

30. Audio compression (data)
Difficulties in lossless compression of audio data
It is difficult to maintain all the data in an audio stream and achieve substantial compression.
First, the vast majority of sound recordings are highly complex, recorded from the real world.
Second, The values of audio samples change very quickly, so generic data compression algorithms don't work well for audio, and strings of consecutive bytes don't generally appear very often.

31. Audio compression (data)
Evaluation criteria of Lossless audio compression
Lossless audio codecs have no quality issues, so the usability can be estimated by
Speed of compression and decompression
Degree of compression
Software and hardware support
Robustness and error correction

32. Audio compression (data)
Lossy audio compression
Lossy audio compression is used in an extremely wide range of applications.
Digitally compressed audio streams are used in many applications, for example:
Video DVDs.
Digital television.
Streaming media on the internet.
Satellite and cable radio.
Increasingly in radio broadcasts.
Lossy compression typically achieves far greater compression than lossless compression (5-20% of the original stream, rather than 50-60% in the Lossless.

33. Audio compression (data)
Lossy audio compression (cont.)
In lossy audio compression, data that can not be perceived by the human auditory system are coded with decreased accuracy or not coded at all.
Most lossy compression reduces perceptual redundancy.
Removing or reducing 'unhearable' sounds may account for a small percentage of bits saved in lossy compression.

34. Audio compression (data)
Lossy audio compression (cont.)
Reducing the number of bits used to code a signal increases the amount of noise in that signal (Recall SNR).
The real key in lossy compression, is to 'hide' the noise generated by the bit savings in areas of the audio stream that cannot be perceived.
This is done by, for instance, using very small numbers of bits to code the high frequencies of most signals.
human ear can only perceive very loud signals in this region
so that softer sounds 'hidden' there simply aren't heard.

35. Audio compression (data)
Lossy audio compression (cont.)
If reducing perceptual redundancy does not achieve sufficient compression for a particular application, it may require further lossy compression.
Depending on the audio source, this still may not produce perceptible differences. Speech for example can be compressed far more than music.
Most lossy compression schemes allow compression parameters to be adjusted to achieve a target rate of data, usually expressed as a bit rate.

36. Audio compression (data)
Lossy audio compression (cont.)
Because data is removed during lossy compression and cannot be recovered by decompression, some people may not prefer lossy compression for archival storage.
People use lossy compression may wish to keep a losslessly compressed archive for other applications.

37. Audio compression (data)
Applications of lossy compression
Due to the nature of lossy algorithms, audio quality suffers when a file is decompressed and recompressed (digital generation loss).
Lossy compression unsuitable for storing the intermediate results in professional audio engineering applications.
Popular with end users (particularly MP3).

38. Audio compression (data)
Evaluation criteria
Usability of lossy audio codecs is determined by:
Perceived audio quality
Compression factor
Speed of compression and decompression
Inherent latency of algorithm (critical for real-time streaming applications; explained later)
Software and hardware support

39. Audio compression (data)
Inherent latency of algorithm
Lossy formats are often used for the distribution of streaming audio, or interactive applications.
In such applications, the data must be decompressed as the data flows.
Not all audio codecs can be used for streaming applications.
For such applications a codec designed to stream data effectively will usually be chosen.

40. Audio compression (data)
Inherent latency of algorithm (cont.)
Latency results from the methods used to encode and decode the data.
Some codecs will analyze a longer segment of the data to optimize efficiency.
Often codecs create segments called a "frame" to create discrete data segments for encoding and decoding.

41. Audio compression (data)
Inherent latency of algorithm (cont.)
Latency refers to the number of samples which must be analysed before a block of audio is processed.
In the minimum case, latency is 0 zero samples (e.g., if the coder/decoder simply reduces the number of bits used to quantize the signal).
Time domain algorithms have low latencies, hence their popularity in speech coding for telephony.