Illustrations

Figures

1.1 Monaural or single-channel listening

1.2 Ideal two-channel stereo loudspeaker and listener placement

1.3 Stereo listening with crosstalk

1.4 Ideal five-channel surround listening placement according to the ITU-R BS.775-1 recommendations (ITU-R, 1994), with the listener equidistant from all five loudspeakers

2.1 The frequency content of a mix at one moment in time

2.2 The time domain (top) and frequency domain (bottom) representations of a 1 kHz sine tone

2.3 The signal path showing the transmission of an audio signal as an electrical signal to a loudspeaker where it is converted to an acoustic signal, modified by a listening room, and finally received by the ear and processed by the auditory system

2.4 A perfect impulse (top, time domain representation) contains all frequencies at equal level (bottom, frequency domain representation)

2.5 The impulse response (top, time domain representation) of an all-pass filter (in this case REAPER’s ReaEQ) and the resulting frequency response (bottom, frequency domain representation))

2.6 The frequency response of a signal mixed with an all-pass filtered version of itself)

2.7 The frequency response of a low-pass filter set to 1000 Hz at three different slopes)

2.8 The frequency response of a high-pass filter set to 1000 Hz at three different slopes)

2.9 The frequency response of a parametric equalizer with a boost of 15 dB at F = 1000 Hz and Q = 2.0)

2.10 The frequency response of a parametric equalizer with a boost of 15 dB at F _c = 1000 Hz and Q = 2.0 (= F _c /bw = 1000/500)

2.11 The frequency response of symmetrical boost/cut equalizer showing a boost of 15 dB overlaid with a cut of − 15 dB at F _c = 1 kHz

2.12 The frequency response of an asymmetric boost/cut equalizer showing a boost of 15 dB and a separate notch cut at F _c = 1000 Hz and Q = 2.0, with the boost and cut applied separately but overlaid on the same plot

2.13 If we apply an asymmetric boost and cut on the same track—in other words, if we sum the asymmetrical boost/cut equalizer for a given center frequency (F _c = 1 kHz in this plot)—the frequency response is far from flat, as we can see in this plot

2.14 A comparison of a high-frequency shelf filter and low-pass filter

2.15 If we record acoustic bass with a bidirectional mic in the same room as saxophone, placing the sax 90° off-axis can help reduce the saxophone spill in the bass microphone

2.16 A screenshot of the software user interface for the Technical Ear Trainer parametric equalization practice module

2.17 A block diagram of the signal path for the Technical Ear Trainer practice module for parametric equalization

3.1 A vectorscope meter showing the stereo image width and correlation of a 1 kHz sine tone panned to the center

3.2 A vectorscope meter showing the stereo image width and correlation of a 1 kHz sine tone panned to the right

3.3 A vectorscope meter showing the stereo image width and correlation of a 1 kHz sine tone with phase reversed (polarity inverted) on one channel of the stereo bus

3.4 A vectorscope meter showing the stereo image width and correlation of a stereo mix

3.5 A vectorscope meter showing the stereo image width and correlation of a stereo mix

3.6 A Blumlein stereo microphone technique uses two coincident figure-8 microphones angled 90 degrees apart

3.7 The top part (A) shows a block diagram of a signal combined with a delayed version of itself, also known as a feedforward comb filter; the bottom part (B) shows the impulse response of the block diagram with a gain of 0.5: a signal (in this case an impulse) plus a delayed version of itself at half the amplitude

3.8 The top part (A) shows a block diagram of a signal combined with a delayed version of itself with the output connected back into the delay, also known as a feedback comb filter; the bottom part (B) shows the impulse response of the block diagram with a gain of 0.5: a signal (in this case an impulse) plus a repeating delayed version of itself where each subsequent delayed output is half the amplitude of the previous one

3.9 A block diagram of an all-pass filter, which is essentially a combination of a feedforward and feedback comb filter

3.10 A block diagram of Manfred Schroeder’s original digital reverberation algorithm, showing four comb filters in parallel that feed two all-pass filters in series, upon which modern conventional reverb algorithms are based

3.11 Impulse responses of three different reverb plug-ins with parameters set as identically as possible: reverb decay time: 2.0 s; predelay time: 0 ms; room type: hall

3.12 A screenshot of the user interface for the spatial trainer

3.13 A block diagram (A) and a mixer signal flow diagram (B) to convert Left and Right stereo signals into Mid (Left + Right) and Side (Left− Right) signals, and subsequent mixing back into Left and Right channels

4.1 The RMS value of a sine wave is always 70.7% of the peak value, which is the same as saying that the RMS value is 3 dB below the peak level

4.2 A square wave has equal peak and RMS levels, so the crest factor is 0

4.3 A pulse wave is similar to a square wave except that we are shortening the amount of time the signal is at its peak level

4.4 The top graph (A) shows the four components of an ADSR (attack, decay, sustain, release) amplitude envelope that describe and generate a synthesized sound; the bottom graph (B) shows an amplitude envelope for an acoustic sound, such as from a string or drum, which can have a relatively fast attack but immediately starts to decay after being struck

4.5 This figure shows a step function, an amplitude-modulated sine wave, that we can use to test the attack and release times of a compressor

4.6 The step response of a compressor showing three different attack and release times: long (A), medium (B), and short (C)

4.7 The same modulated 40-Hz sine tone through a commercially available analog compressor with an attack time of approximately 50 ms and a release time of 200 ms

4.8 From an audio signal (A) sent to the input of a compressor, a gain function (B) is derived based on compressor parameters and signal level; the resulting audio signal output (C) from the compressor is the input signal with the gain function applied to it

4.9 This figure shows the step response of an expander for three different attack and release times: long (A), medium (B), and short (C)

4.10 From an audio signal (A) sent to the input of an expander, a gain function (B) is derived based on expander parameters and signal level; the resulting audio signal output (C) from the expander is the input signal with the gain function applied to it

4.11 A screenshot of the software user interface for the Technical Ear Trainer practice module for dynamic range compression

5.1 A sine wave at 1 kHz

5.2 A sine wave with crossover distortion

5.3 A sine wave at 1 kHz that has been hard clipped

5.4 A sine wave at 1 kHz that has been soft clipped or overdriven

5.5 A sine wave at 1 kHz that has been quantized with 3 bits, giving 8 (or 2 3) steps

6.1 A typical view of a waveform in a digital editor with the edit point marker that indicates where the edit point will occur and the audio will cross-fade into a new take

6.2 The software module presented here re-creates the process of auditioning a sound clip up to a predefined point and matching that end point in a second sound clip

6.3 Source and destination waveform timelines are shown here in block form along with an example of how a set of takes (source) might fit together to form a complete performance (destination)

6.4 Clips of a music recording of four different lengths: 825 ms, 850 ms, 875 ms, and 900 ms

6.5 A screenshot of the training software

7.1 I encourage you to use this template as a guide for the graphical analysis of a sound image, to visualize the perceived locations of sound images within a sound recording

7.2 This is an example of a graphical analysis of a stereo image of a jazz piano trio recording

Table

2.1 The complete list of frequencies (in Hz) shown with octave frequencies in bold