Abstract: A system and method of improving the efficiency in the power consumption of an audio system. In essence, the technique is to adjust the power delivered from the power supply to the analog section, such as the power amplifier, in response to the volume level indicated by the volume control module and/or in response to the detected characteristic of the input audio signal. Thus, in this manner, the analog section is operated in a manner that is related to the level of the signal it is processing. Additionally, the system and method also relate to a technique of adjusting the dynamic ranges of the digital signal and the analog signal to improve the overall dynamic range of the system without needing to consume additional power.

Abstract: In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.

Abstract: A method for providing an interface to a processing engine that utilizes intelligent audio mixing techniques may include receiving a request to change a perceptual location of an audio source within an audio mixture from a current perceptual location relative to a listener to a new perceptual location relative to the listener. The audio mixture may include at least two audio sources. The method may also include generating one or more control signals that are configured to cause the processing engine to change the perceptual location of the audio source from the current perceptual location to the new perceptual location via separate foreground processing and background processing. The method may also include providing the one or more control signals to the processing engine.

Abstract: This disclosure describes audio techniques that exploit a prioritization scheme of audio streams of a software application, such as a video game that execute on a mobile device. The priority of the audio streams are defined by a content creator of the application. In addition, priority of an additional audio stream not associated with the application (such as an audio alert) is also defined. A mobile device includes a processor that executes the application and an audio decoding unit that receives a plurality of prioritized audio streams of an application executing on a mobile device, receives an additional prioritized audio stream not associated with the application, and combines a subset of the prioritized audio streams associated with the application and the additional prioritized audio streams according to priority to form a common audio stream.

Abstract: This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.

Abstract: This disclosure describes techniques that make use of a waveform fetch unit that operates to retrieve waveform samples on behalf of each of a plurality of hardware processing elements that operate simultaneously to service various audio synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises receiving a request for a waveform sample from an audio processing element, and servicing the request by calculating a waveform sample number for the requested waveform sample based on a phase increment contained in the request and an audio synthesis parameter control word associated with the requested waveform sample, retrieving the waveform sample from a local cache using the waveform sample number, and sending the retrieved waveform sample to the requesting audio processing element.

Abstract: A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

Abstract: This disclosure describes techniques that make use of a summing buffer that receives waveform samples from audio processing elements, and sums and stores the waveform sums for a given frame. In one example, a method comprises summing a waveform sample received from an audio processing element to produce a waveform sum associated with a first audio frame, storing the waveform sum in a memory, wherein the memory is logically partitioned into a plurality of memory blocks, and locking memory blocks containing the waveform sum associated with the first audio frame, transferring contents of locked memory blocks to an external processor, unlocking a memory block after contents of the memory block have been transferred to the external processor, and storing a waveform sum associated with a second audio frame within the unlocked memory block concurrently with transferring contents of remaining locked memory blocks associated with the first audio frame.

Abstract: Generating a digital waveform for a Musical Instrument Digital Interface (MIDI) voice using a set of machine-code instructions that is specialized for the generation of digital waveforms for MIDI voices. For example, a processor may execute a software program that generates a digital waveform for a MIDI voice. The instructions of the software program may be machine code instructions from an instruction set that is specialized for the generation of digital waveforms for MIDI voices.

Abstract: A system and method of improving the efficiency in the power consumption of an audio system. In essence, the technique is to adjust the power delivered from the power supply to the analog section, such as the power amplifier, in response to the volume level indicated by the volume control module and/or in response to the detected characteristic of the input audio signal. Thus, in this manner, the analog section is operated in a manner that is related to the level of the signal it is processing. Additionally, the system and method also relate to a technique of adjusting the dynamic ranges of the digital signal and the analog signal to improve the overall dynamic range of the system without needing to consume additional power.

Abstract: A method for providing an interface to a processing engine that utilizes intelligent audio mixing techniques may include receiving a request to change a perceptual location of an audio source within an audio mixture from a current perceptual location relative to a listener to a new perceptual location relative to the listener. The audio mixture may include at least two audio sources. The method may also include generating one or more control signals that are configured to cause the processing engine to change the perceptual location of the audio source from the current perceptual location to the new perceptual location via separate foreground processing and background processing. The method may also include providing the one or more control signals to the processing engine.

Abstract: In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.

Abstract: Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter.

Abstract: A technique for canceling acoustic crosstalk is provided including a pre-processing filter and a crosstalk cancellation device. The pre-processing filter may be configured to obtain first and second channel signals and compensate or adjust the first and/or second channel signals for anticipated subsequent stage distortion by the crosstalk cancellation device. The crosstalk cancellation device maybe configured to receive the compensated first and second channel signals from the pre-processing filter. The crosstalk cancellation device then modifies the first channel signal to cancel anticipated acoustic crosstalk from the second channel signal, and modifies the second channel signal to cancel acoustic crosstalk from the first channel signal. The modified first channel signal is then transmitted over a first speaker and the modified second channel signal is transmitted over a second speaker.

Abstract: A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

Abstract: Generating a digital waveform for a Musical Instrument Digital Interface (MIDI) voice using a set of machine-code instructions that is specialized for the generation of digital waveforms for MIDI voices. For example, a processor may execute a software program that generates a digital waveform for a MIDI voice. The instructions of the software program may be machine code instructions from an instruction set that is specialized for the generation of digital waveforms for MIDI voices.

Abstract: This disclosure describes techniques that make use of a waveform fetch unit that operates to retrieve waveform samples on behalf of each of a plurality of hardware processing elements that operate simultaneously to service various audio synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises receiving a request for a waveform sample from an audio processing element, and servicing the request by calculating a waveform sample number for the requested waveform sample based on a phase increment contained in the request and an audio synthesis parameter control word associated with the requested waveform sample, retrieving the waveform sample from a local cache using the waveform sample number, and sending the retrieved waveform sample to the requesting audio processing element.

Abstract: This disclosure describes techniques that make use of a summing buffer that receives waveform samples from audio processing elements, and sums and stores the waveform sums for a given frame. In one example, a method comprises summing a waveform sample received from an audio processing element to produce a waveform sum associated with a first audio frame, storing the waveform sum in a memory, wherein the memory is logically partitioned into a plurality of memory blocks, and locking memory blocks containing the waveform sum associated with the first audio frame, transferring contents of locked memory blocks to an external processor, unlocking a memory block after contents of the memory block have been transferred to the external processor, and storing a waveform sum associated with a second audio frame within the unlocked memory block concurrently with transferring contents of remaining locked memory blocks associated with the first audio frame.

Abstract: This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.

Abstract: Techniques are described for sampling rate conversion in the digital domain by up-sampling and down-sampling a digital signal according to a selected intermediate sampling frequency. A prototype anti-aliasing filter that has a bandwidth with multiple factors is stored in memory. The techniques include selecting an intermediate sampling frequency to be an integer multiple of a desired output sampling frequency of a digital signal based on the factors of the prototype filter, and selecting a down-sampling factor to be the same integer associated with the selected intermediate sampling frequency. A filter generator generates an anti-aliasing filter for the selected down-sampling factor based on the prototype filter.