The Best DSP eBooks of All Time

Designing Audio Effect Plugins in C++ presents everything you need to know about digital signal processing in an accessible way. Not just another theory-heavy digital signal processing book, nor another dull build-a-generic-database programming book, this book includes fully worked, downloadable code for dozens of professional audio effect plugins and practically presented algorithms. Sections include the basics of audio signal processing, the anatomy of a plugin, AAX, AU and VST3 programming guides; implementation details; and actual projects and code. More than 50 fully coded C++ audio signal-processing objects are included.

Start with an intuitive and practical introduction to the digital signal processing (DSP) theory behind audio plug-ins, and quickly move on to plugin implementation, gain knowledge of algorithms on classical, virtual analog, and wave digital filters, delay, reverb, modulated effects, dynamics processing, pitch shifting, nonlinear processing, sample rate conversion and more. You will then be ready to design and implement your own unique plugins on any platform and within almost any host program. This new edition is fully updated and improved and presents a plugin core that allows readers to move freely between application programming interfaces and platforms. Readers are expected to have some knowledge of C++ and high school math.

This eBook provides the audio enthusiast with an easy-to-follow step-by-step guide for designing a custom digital filter that corrects the frequency and timing response of your loudspeakers in your listening environment so that the music arriving at your ears matches as closely as possible to the content on the recording. Industry guidelines, spanning over 40 years of evolution, are referenced throughout the eBook, providing the recommended target responses for accurate sound reproduction.

Correcting the measured response to known target responses is, in effect, matching the acoustic output signal arriving at your ears to the digital audio stored in a media file. The target responses can also be tailored to your own preferences. The chapter on the acoustics and psychoacoustics of room correction explains why we hear what we hear in small-room acoustics and what our ears care about most when it comes to accurate sound reproduction. There are individual chapters on industry target responses for frequency and step response, group delay, energy time curve, reverb time, polar response and interaural coherence coefficient (i.e., imaging). Advanced sections of the eBook show you how to design and implement digital crossovers, driver linearization, and driver time alignment; how to compensate for brickwall antialiasing filters; and how to use the Beamforming quasi-anechoic measurement technique. All of these sections are designed to incrementally improve your sound reproduction system’s accuracy. The last section of the eBook shows you how to perform loopback measurements to verify the design and operation of your digital correction filter, not only at the listening position, but across a wide listening area as well. Whether your sound reproduction system is a computer desktop speaker system, stereo audiophile system, home theater multi-channel system, mains and sub-woofer system, DIY audio system, or car sound system, the results are audible, measurable, and repeatable. To benefit the most from this step-by-step guide, you will require a computer, a calibrated measurement microphone, an Analog to Digital (ADC)/Digital to Analog (DAC) converter, and two commercial software packages: Acourate DSP Audio Toolbox and JRiver Media Center. This eBook contains several color-coded images, so viewing on a color screen is required. With over 260 color screen captures of the software in action, plus 200 links to industry reference material and knowledge, this eBook is the definitive guide to Accurate Sound Reproduction Using Digital Signal Processing (DSP).

Amazon.com’s Top-Selling DSP Book for Seven Straight Years―Now Fully Updated! Understanding Digital Signal Processing, Third Edition, is quite simply the best resource for engineers and other technical professionals who want to master and apply today’s latest DSP techniques. Richard G. Lyons has updated and expanded his best-selling second edition to reflect the newest technologies, building on the exceptionally readable coverage that made it the favorite of DSP professionals worldwide. He has also added hands-on problems to every chapter, giving students even more of the practical experience they need to succeed.

Comprehensive in scope and clear in approach, this book achieves the perfect balance between theory and practice, keeps math at a tolerable level, and makes DSP exceptionally accessible to beginners without ever oversimplifying it. Readers can thoroughly grasp the basics and quickly move on to more sophisticated techniques. This edition adds extensive new coverage of FIR and IIR filter analysis techniques, digital differentiators, integrators, and matched filters. Lyons has significantly updated and expanded his discussions of multirate processing techniques, which are crucial to modern wireless and satellite communications. He also presents nearly twice as many DSP Tricks as in the second edition―including techniques even seasoned DSP professionals may have overlooked. Coverage includes. New homework problems that deepen your understanding and help you apply what you’ve learnedPractical, day-to-day DSP implementations and problem-solving throughoutUseful new guidance on generalized digital networks, including discrete differentiators, integrators, and matched filtersClear descriptions of statistical measures of signals, variance reduction by averaging, and real-world signal-to-noise ratio (SNR) computationA significantly expanded chapter on sample rate conversion (multirate systems) and associated filtering techniquesNew guidance on implementing fast convolution, IIR filter scaling, and moreEnhanced coverage of analyzing digital filter behavior and performance for diverse communications and biomedical applicationsDiscrete sequences/systems, periodic sampling, DFT, FFT, finite/infinite impulse response filters, quadrature (I/Q) processing, discrete Hilbert transforms, binary number formats, and much more.