Cirrus Logic Inc.

Silicon Laboratories Inc.

St-ericsson Sa

Dylan Alexander Hester

Bruce Eliot Duewer

Shafagh Kamkar

Augusto Manuel Marques

John L Melanson

Donald A Kerth

G Tyson Tuttle

Scott Thomas Haban

Ruifeng Sun

James P Maligeorgos

Krishnan Subramoniam

Rajendra Datar

Anand Venkitachalam

Bala Vishnu Shankar Rao

Daniel John Allen

John Christopher Tucker

John James Paulos

Emmanuel A Marchais

Russell Croman

Cory Jay Peterson

Lysander Lim

Xue-Mei Gong

Steven Green

Mark Alexander

Eric Clay Gaalaas

Jason Lee Wardlaw

Joe Welser

Christophe Jean-Francois Amadi

Brian Frank Bounds

Tarun Soni

A single-bit audio stream can be converted to a modified single-bit audio stream with a constant edge rate while maintaining a modulation index of the original audio stream using direct mapping. With direct mapping, a pre-filter bank may be combined with a multi-bit symbol mapper to select symbols for the modified audio stream with a constant edge rate per symbol and the same modulation index as the original audio stream. The output of the pre-filter bank may be an audio stream with no consecutive full-scale symbols. Using the output of the pre-filter bank, a multi-bit symbol mapper may use the symbol selector to output a symbol with a constant edge rate per symbol and the same modulation index as the original signal. The symbols may be converted to an analog signal for reproduction of audio content using a transducer.

Direct mapping

Converting a single-bit audio stream to a stream of symbols with a constant edge rate

An encoding for data in an audio data stream may be indicated in the data stream using a footer stored in low-order bits of data frames in the audio data stream. When the audio data stream may include either Pulse Code Modulation (PCM) or Direct Stream Digital (DSD) data, PCM data may be marked with a footer to indicate the encoding as PCM. The footer may be a fixed value, an alternating fixed value, a predetermined sequence of values, or a value computed based on the PCM data. Examples of computed values for the footer marker may include an error code, an error correction code (ECC), and a scrambled code.

Pulse code modulation (PCM) data-marking

Audio content in a single-bit audio stream can be reproduced at a transducer by mapping the single-bit audio stream to symbols in a multi-bit audio stream. Volume control may be implemented, in part, in the digital domain and, in part, in the analog domain. In the digital domain, when converting the single-bit audio stream to a plurality of symbols, the plurality of symbols is selected based, at least in part, on audio content of the single-bit audio stream and a desired volume level. In the analog domain, when converting an analog current signal output from a current-steering DAC processing the plurality of symbols to an analog voltage signal, an analog gain value may be selected based, at least in part, on the desired volume level.

Single-bit volume control

An audio processing circuit may have a first path for processing multi-bit audio signals in parallel with a second path for processing single-bit audio signals. The parallel paths may share a common input node for receiving audio data and a common output node for reproducing audio at a transducer. Each path may have a volume control for adjusting an output of the path. The audio processing circuit may determine a type of an audio signal received at the input. The path not corresponding to the detected type of the audio signal is muted, and the path corresponding to the detected type of audio signal is unmuted.

Auto-mute audio processing

Converting a single-bit audio stream to a single-bit audio stream with a constant edge rate

Droop caused by a filter may be compensated by applying a pre-filter to the audio signal that cancels out, at least in part, the droop caused by the filter. The pre-filter may implement magnitude compensation that causes an approximately flat passband response when the pre-filtered signal is passed through the filter. The pre-filter may be applied to one-bit wide data streams, such as high-fidelity direct stream digital (DSD) audio data or other one-bit wide data such as pulse-density modulation (PDM) encoded data. The pre-filtering and filtering may be implemented in components of an audio processor, such as in a digital-to-analog converter (DAC). The pre-filtering may include upsampling the one-bit wide data to form symbols and substituting an eighth bit of the symbol with an inverted version of an earlier-received bit.

Magnitude compensation technique for processing single-bit wide data

A data encoding may be determined by examining the data itself. By examining the data, such as by identifying characteristics of the data, the data encoding may be identified and the substantive information extracted from the data by decoding the data according to the identified data encoding. The data encoding may be identified without extra control or header information separate from the encoded data itself. The identification of data encoding may be determined by a plurality of detectors examining different characteristics of the data and determining an encoding of the data based on each of those individual characteristics. The output of the detectors may be collected and used to decide the encoding of the data. One example application involves determining if audio data is DSD- or PCM-encoded audio data by examining only the data itself.

Data encoding detection

An audio processor may process dissimilarly formatted digital audio signals and produce an analog output for reproduction by a transducer, such as a speaker. The audio processor may be, for example, a digital-to-analog converter (DAC) with multiple digital inputs and a single analog output. The audio processor may also include other components, such as processing components, filters, equalizers, amplifiers, and the like. The audio processor may process audio, such as high fidelity DSD audio and lower fidelity PCM audio, and combine the differently formatted data into a single output. During processing, the audio processor may manipulate the different audio signals, such as by ducking a music playback to allow a notification sound to be played.

Mixing of single-bit and multi-bit audio signals for simultaneous output

The behavior of a NOS DAC and an analog filter may be emulated by electronic components of an integrated circuit (IC) by upsampling data and applying a digital filter to the upsampled data. For example, the IC may include a zero-order-hold circuit that upsamples data from a first input sample rate to a second, higher input rate. The upsampled data may be passed to an Asynchronous Sample Rate Converter (ASRC) that performs further upsampling (e.g., from 8*Fs-64*Fs). The upsampled data may be passed to a digital low pass filter. The digital low pass filter may emulate, for example, a response of a fifth order Butterworth analog filter to mimic the effect of analog processing. The IC may integrate the upsampling circuit, the low pass digital filter, a digital-to-analog converter (DAC) and an amplifier to provide an audio solution for playing high-fidelity music in a mobile device.

Over-sampling digital processing path that emulates Nyquist rate (non-oversampling) audio conversion

A digital volume control may be implemented in a digital-to-analog controller (DAC) when an output rate of the DAC is higher than the input rate. The upsampling conversion process from the digital input to analog output may be controlled to adjust a volume of an output signal produced from the digital signals. The frames produced by the upsampling conversion process may include a fraction of scheduling blocks filled with the input data bit, where the fraction is based on a desired volume. The generated frames are provided to a finite impulse response (FIR) filter that produces an analog signal with a magnitude proportional to the determined fraction of scheduling blocks.

Digital volume control

A DC servo loop may track DC offset changes of an input signal and apply feedback to amplifiers to adjust a DC offset of the input signal. The DC servo loop may include digital loop tracking and analog loop tracking components. The digital loop tracking components may track small changes in the DC offset. When the DC offset exceeds a certain threshold, analog loop tracking may be activated to apply feedback to the amplifiers to adjust the DC offset. The adjustments to the DC offset may be delayed until an amplitude of the input signal exceeds a threshold to reduce contribution to noise in the input signal.

Feedback loop for direct current (DC) offset removal

A demodulator can include first data and clock pads to couple the demodulator to a host device via a first bus, and second data and clock pads to couple the demodulator to a radio frequency (RF) tuner via a second bus. The device may further include passthrough logic to couple host data and a host clock from the first bus to the second bus and to couple tuner data from the second bus to the first bus during a passthrough mode. During this mode, however, the two buses may remain electrically decoupled. When the passthrough mode is disabled, the RF tuner is thus shielded from noise present on the first bus.

Controlling passthrough of communication between multiple buses

Controlling passthrough of communications between multiple buses

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Austin, TX, United States of America

Dylan Alexander Hester