Abstract: A device renders enhanced audio based on properties of an audio rendering system. For example, audio rendering system information of an audio rendering system associated with the device is used to determine an audio enhancement optimized for the audio rendering system. The OS of the device (e.g., a mobile phone) is queried to determine the audio rendering system information. The audio rendering system information is used to determine an audio enhancement, such as by querying a database storing associations between rendering system information and audio enhancements. The audio enhancement may include, for example, one or more types of audio processing (e.g., subband spatial processing, crosstalk processing, etc.) and particular parameters for the audio processing. The audio enhancement is applied to an audio signal to generate an enhanced audio signal that is provided to the audio rendering system.

Abstract: A device renders enhanced audio based on properties of an audio rendering system. For example, audio rendering system information of an audio rendering system associated with the device is used to determine an audio enhancement optimized for the audio rendering system. The OS of the device (e.g., a mobile phone) is queried to determine the audio rendering system information. The audio rendering system information is used to determine an audio enhancement, such as by querying a database storing associations between rendering system information and audio enhancements. The audio enhancement may include, for example, one or more types of audio processing (e.g., subband spatial processing, crosstalk processing, etc.) and particular parameters for the audio processing. The audio enhancement is applied to an audio signal to generate an enhanced audio signal that is provided to the audio rendering system.

Abstract: A device renders enhanced audio based on properties of an audio rendering system. For example, audio rendering system information of an audio rendering system associated with the device is used to determine an audio enhancement optimized for the audio rendering system. The OS of the device (e.g., a mobile phone) is queried to determine the audio rendering system information. The audio rendering system information is used to determine an audio enhancement, such as by querying a database storing associations between rendering system information and audio enhancements. The audio enhancement may include, for example, one or more types of audio processing (e.g., subband spatial processing, crosstalk processing, etc.) and particular parameters for the audio processing. The audio enhancement is applied to an audio signal to generate an enhanced audio signal that is provided to the audio rendering system.

Abstract: A device renders enhanced audio based on properties of an audio rendering system. For example, audio rendering system information of an audio rendering system associated with the device is used to determine an audio enhancement optimized for the audio rendering system. The OS of the device (e.g., a mobile phone) is queried to determine the audio rendering system information. The audio rendering system information is used to determine an audio enhancement, such as by querying a database storing associations between rendering system information and audio enhancements. The audio enhancement may include, for example, one or more types of audio processing (e.g., subband spatial processing, crosstalk processing, etc.) and particular parameters for the audio processing. The audio enhancement is applied to an audio signal to generate an enhanced audio signal that is provided to the audio rendering system.

Abstract: A device renders enhanced audio based on properties of an audio rendering system. For example, audio rendering system information of an audio rendering system associated with the device is used to determine an audio enhancement optimized for the audio rendering system. The OS of the device (e.g., a mobile phone) is queried to determine the audio rendering system information. The audio rendering system information is used to determine an audio enhancement, such as by querying a database storing associations between rendering system information and audio enhancements. The audio enhancement may include, for example, one or more types of audio processing (e.g., subband spatial processing, crosstalk processing, etc.) and particular parameters for the audio processing. The audio enhancement is applied to an audio signal to generate an enhanced audio signal that is provided to the audio rendering system.

Abstract: Apparatus and methods are provided for encoding video data in a manner which significantly reduces the computation performed by the video encoder and the video decoder without suffering any degradation in the perceived quality of the compressed video data. In particular, apparatus and methods are provided for determining which blocks might be zeroed out after quantization. This determination is performed after motion estimation, the classification of the frame as either an I frame, P frame, or a B frame, and the determination of a quantization step size (QP) for the block, but before DCT. If a given block is determined to be a “zero” quantized block, then the DCT, quantization, zig-zag scan and variable length coding steps are omitted, and a variable length code output is provided indicating that the block B is a “zero” quantized block.

Abstract: The present invention is directed to a voice tone discriminator for distinguishing between call progress tones and voice. The voice tone discriminator is useful in various applications involving the ability to automatically charge a telephone user based on the exact time the user starts speaking. According to another aspect of the present invention, a prediction algorithm may be implemented for distinguishing voice and tone based on the fact that tones are more accurately modeled with a linear filter than voice signals. Thus, a low order filter or predictor may accurately model redundancies in tones but not in voice. A normalized error between an original and a predicted signal may be used to distinguish voice from tones. Voice may be detected when the error is above a preset threshold for a time greater than a preset fixed duration.

Abstract: The present invention pertains to an apparatus (and corresponding method) including both an echo cancellation module and packet loss concealment module for the processing of audio samples sent within packet data networks. The echo cancellation module serves the role of reducing echoes that may have been introduced in the audio samples. The packet loss concealment module serves the role of concealing loss of audio samples that may have occurred in the received audio samples. The echo cancellation module includes a buffer for storing recently received audio samples. This buffer provides historical data for use by both the echo cancellation module and the packet loss concealment module in performing their respective functions. The use of one data buffer (instead of separate buffers for the echo cancellation module and the packet loss concealment module) provides a resource-efficient solution, allowing the invention to be implemented in a cost-effective and size-efficient manner.

Abstract: Apparatus and methods are provided for encoding video data in a manner which significantly reduces the computation performed by the video encoder and the video decoder without suffering any degradation in the perceived quality of the compressed video data. In particular, apparatus and methods are provided for determining which blocks might be zeroed out after quantization. This determination is performed after motion estimation, the classification of the frame as either an I frame, P frame, or a B frame, and the determination of a quantization step size (QP) for the block, but before DCT. If a given block is determined to be a “zero” quantized block, then the DCT, quantization, zig-zag scan and variable length coding steps are omitted, and a variable length code output is provided indicating that the block B is a “zero” quantized block.

Abstract: A pulse train of primary pulses is estimated from an inverse LPC analysis of a frame of voiced speech. From this estimated pulse train a pole-zero filter is estimated. The estimated pulse train is used to excite the estimated pole-zero filter to produce a synthesized speech signal. The synthesized speech signal is compared to the original frame of speech to determine the error in the original speech signal. Both the pulse amplitude and filter are adjusted to compensate for the error and another synthesized speech signal is produced. The process may be repeated until the synthesized speech signal and original speech signal converge.