Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: An application and audio driver communicate audio data via a ring buffer using a system interface. An application reads or writes data from the buffer in a unit of data. Positions within the buffer correspond to a particular time. To write data to the buffer at a particular position in the buffer that corresponds to a particular time, an application is awaken a period of time (“client-side offset”) before the driver reaches the position. The period of time is computed based on a variety factors. One factor is the amount of time the system interface assumes an application uses to write the unit of data to the buffer. For operations that entail the application inserting data into the buffer, the application may specify a weight value to apply to the factor to reduce it and the latency period.

Abstract: Some embodiments of the invention provide a computer system for processing an audio track. This system includes at least on DSP for processing the audio track. It also includes an application for editing the audio track. To process audio data in a first interval of the audio track, the application first asks and obtains from the DSP an impulse response parameter related to the DSP's processing of audio data. From the received impulse response parameter, the application identifies a second audio track interval that is before the first interval. To process audio data in the first interval, the application then directs the DSP to process audio data within the first and second intervals.

Abstract: An application and audio driver communicate audio data via a ring buffer using a system interface. An application reads or writes data from the buffer in a unit of data. Positions within the buffer correspond to a particular time. To write data to the buffer at a particular position in the buffer that corresponds to a particular time, an application is awaken a period of time (“client-side offset”) before the driver reaches the position. The period of time is computed based on a variety factors. One factor is the amount of time the system interface assumes an application uses to write the unit of data to the buffer. For operations that entail the application inserting data into the buffer, the application may specify a weight value to apply to the factor to reduce it and the latency period.

Abstract: An application and audio driver communicate audio data via a ring buffer using a system interface. An application reads or writes data from the buffer in a unit of data. Positions within the buffer correspond to a particular time. To write data to the buffer at a particular position in the buffer that corresponds to a particular time, an application is awaken a period of time (“client-side offset”) before the driver reaches the position. The period of time is computed based on a variety factors. One factor is the amount of time the system interface assumes an application uses to write the unit of data to the buffer. For operations that entail the application inserting data into the buffer, the application may specify a weight value to apply to the factor to reduce it and the latency period.

Abstract: Techniques for mixing multiple input channel signals into multiple output channel signals are provided. A graphical user interface (GUI), which includes multiple indicators, is displayed. The input channel signals are mixed to produce multiple output channel signals. The mixing is performed based on the distance between the indicators' positions in the GUI. According to one embodiment of the invention, the mixing is also performed based on the angle formed between the indicators. Thus, the extent to which an input channel signal is carried by an output channel signal is, in one embodiment of the invention, a function of both the distance between the indicators and an angle formed by the indicators in the GUI.

Abstract: Techniques for mixing multiple input channel signals into multiple output channel signals are provided. A graphical user interface (GUI), which includes multiple indicators, is displayed. The input channel signals are mixed to produce multiple output channel signals. The mixing is performed based on the distance between the indicators' positions in the GUI. According to one embodiment of the invention, the mixing is also performed based on the angle formed between the indicators. Thus, the extent to which an input channel signal is carried by an output channel signal is, in one embodiment of the invention, a function of both the distance between the indicators and an angle formed by the indicators in the GUI.

Abstract: Some embodiments of the invention provide a computer system for processing an audio track. This system includes at least one DSP for processing the audio track. It also includes an application for editing the audio track. To process audio data in a first interval of the audio track, the application first asks and obtains from the DSP an impulse response parameter related to the DSP's processing of audio data. From the received impulse response parameter, the application identifies a second audio-track interval that is before the first interval. To process audio data in the first interval, the application then directs the DSP to process audio data within the first and second intervals.

Abstract: Some embodiments of the invention provide a computer system for processing an audio track. This system includes at least on DSP for processing the audio track. It also includes an application for editing the audio track. To process audio data in a first interval of the audio track, the application first asks and obtains from the DSP an impulse response parameter related to the DSP's processing of audio data. From the received impulse response parameter, the application identifies a second audio track interval that is before the first interval. To process audio data in the first interval, the application then directs the DSP to process audio data within the first and second intervals.