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An efficient method to linearize the switching (power) stage of open-loop class D amplifiers is presented. This technique has been successfully applied to an open-loop fully-digital PWM class D amplifier designed in a 40 nm CMOS process leading to nearly 15 dB improvement in the Total Harmonic Distortion (THD). Simulated open-loop class D amplifier performance resulted to 105 dBA Signal-to-Noise Ratio (SNR), and 1W output power over 8 Ohm with 90% power efficiency and 0.014% THD.
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A precise SPICE model for miniature (MT) triode tubes of high-µ 12AX7 and medium-µ 12AU7 is proposed, based on the physical analysis of the measurement results. Comparing the characteristics between these tubes, the grid current at lower plate voltage and positive grid bias condition is modeled successfully with novel perveance parameters for the first time, though it was known that the perveance depends on both grid and plate bias. It is shown that the modulation factor of the space charge for the MT triodes is different from the other classic tubes. The model is implemented in LTspice to result in a good replication for a variation of three-order magnitude of grid current and cathode current.
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With this paper we present a practically relevant investigation of current, commercially available MEMS microphones (Micro-ElectroMechanical Systems). We compared the static noise floor exhibited by single and various parallel MEMS microphone configurations and a conventional and commonly used electret capsule, as well as the directivity patterns of selected configurations. The results suggest that while current types are exhibiting an already acceptable static noise floor, a direct parallel circuit of MEMS microphones allows further reductions of the noise floor close to the theoretical value of 3 dB SPL per doubling of number of microphones while maintaining omnidirectionality below 5 kHz.
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“Class-D” switching amplifiers are considered to be the most efficient amplifiers available on the market. However, designers must deal with supply rail, and radio frequency interference, as well as the need to switch power devices at high frequencies. Because of these, and other problems, not everyone wishes to use switching based technologies for their amplifiers. Unfortunately, linear amplifiers are significantly more inefficient than switching amplifiers, under sine wave testing. However real audio signals spend much more time at low amplitudes than a sine wave. By changing the switch points for “Class-G” or “Class-H” they can have efficiencies that rival “Class-D” amplifiers producing the same output. The paper develops optimum switch points for both single and multiple switching points, with respect to the expected amplitude distribution of the audio.
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The MPEG-4 AAC Enhanced Low Delay (AAC-ELD) coder is well established in high quality communication applications, such as Apple’s FaceTime, as well as in professional live broadcasting. Both applications require high interactivity, which typically demands an algorithmic codec delay between 15 ms and 35 ms. Recently, MPEG finalized a new delay-reduced mode for AAC-ELD featuring only a fraction of the regular algorithmic delay. This mode operates virtually at higher sampling rates while maintaining standard sampling rates for I/O. Supporting this feature, AAC-ELD can address even more delay critical applications, like wireless microphones or headsets for TV. In this paper main details of the delay-reduced mode of AAC-ELD are presented and application scenarios are outlined. Audio quality aspects are discussed and compared against other codecs with a delay below 10 ms.
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High rate Direct Stream Digital (DSD) is emerging as a format of choice for distribution of high-definition audio content. However, real-time encoding of such streams requires considerable computing resources due to their high sampling rate, constraining implementations to hardware based platforms. In this paper we disclose a new modulator topology allowing for reduction in computational load and making real-time high rate DSD encoding suitable for software based implementation on off-the-shelf Digital Signal Processors (DSPs). We first present the architecture of the proposed modulator and then show results from a practical real-time implementation.
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Bandwidth extension methods, such as spectral band replication (SBR), are often used in low-bit-rate codecs. They allow transmitting only a relatively narrow low-frequency region alongside with parametric information about the higher bands. The signal for the higher bands is obtained by simply copying it from the transmitted low-frequency region. The copied-up signal is processed by multiplying the magnitude spectrum with suitable gains based on the transmitted parameters to obtain a similar magnitude spectrum as that of the original signal. However, the phase spectrum of the copied-up signal is typically not processed but is directly used. In this paper we describe what are the perceptual consequences of using directly the copied-up phase spectrum. Based on the observed effects, two metrics for detecting the perceptually most significant effects are proposed. Based on these, methods how to correct the phase spectrum are proposed as well as strategies for minimizing the amount of transmitted additional parameter values for performing the correction. Finally, the results of formal listening tests are presented.
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AC-4 is a state of the art audio codec standardized in ETSI (TS103 190 and TS103 190-2) and the TS103 190 is part of the DVB toolbox (TS101 154). AC-4 is an audio codec designed to address the current and future needs of video and audio entertainment services including broadcast and Internet streaming. As such, it incorporates a number of features beyond the traditional audio coding algorithms, such as capabilities to support immersive and personalized audio, support for advanced loudness management, video-frame synchronous coding, dialogue enhancement, etc. This paper will outline the thinking behind the design of the AC-4 codec, explain the different coding tools used, the systemic features included, and give an overview of performance and applications.
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Sinusoidal modeling is one of the most common techniques for general purpose audio synthesis and analysis. Owing to the ever increasing amount of available computational resources, nowadays practically all types of sounds can be constructed up to a certain degree of perceptual accuracy. However, the method is computationally expensive and can for some cases, particularly for transient signals, still exceed the available computational resources. In this work methods derived from the realm of machine learning are exploited to provide a simple and efficient means to estimate the achievable reconstruction quality. The peculiarities of common classes of musical instruments are discussed and finally, the existing metrics are extended by information on the signal's phase propagation to allow for more accurate estimations.
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Frequency response equalization is often performed to improve audio reproduction. Variations from the target system response due to playback equipment or room acoustics can result in perceptible timbre distortion. In the first part of this paper we describe experiments conducted to determine the audibility of artificially introduced spectral dips. In particular, we measured notch depth detection threshold (independent variable) with respect to notch center frequency and Q-factor (independent variables). Listening tests were administered to 10 listeners in a small listening room and a screening room (small cinema with approximately 100 seats). Pink noise was used as the stimulus as it is perceptually flat (with roughly 3 dB/octave spectral tilt with frequency) and is known to be a reliable and discriminating signal for performing timbre judgments. The listeners gave consistent notch depth results with low variability around the mean value. The notch audibility data was then used to develop multiple candidate algorithms that generate equalization curves designed to perceptually match a desired target response, while minimizing the equalization gain applied. Informal subjective results validated the performance of the final algorithm.
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