Microsoft Technology Licensing, LLC

Other

Microsoft Corporation

Wei-Ge Chen

Naveen Thumpudi

Sanjeev Mehrotra

Chao He

Kazuhito Koishida

Ming-Chieh Lee

Qin Li

Zhengyou Zhang

Serge Smirnov

Chris Messer

Tian Wang

Hosam A Khalil

Xiaoqin Sun

Henrique Sarmento Malvar

James David Johnston

Hosam Adel Khalil

Mu Han

Sudheer Sirivara

Timothy E Onders

Raymond Cheng

Christopher Stephen Messer

Yoomi Hur

Jaemo Yang

Sanjeev Mehrohtra

An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Bitstream syntax for multi-process audio decoding

An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source. In the header reduction technique, the audio encoder selectively modifies the quantization step size of zeroed quantization bands so as to encode in fewer frame header bits.

Quality improvement techniques in an audio encoder

An encoder performs context-adaptive arithmetic encoding of transform coefficient data. For example, an encoder switches between coding of direct levels of quantized transform coefficient data and run-level coding of run lengths and levels of quantized transform coefficient data. The encoder can determine when to switch between coding modes based on a pre-determined switch point or by counting consecutive coefficients having a predominant value (e.g., zero). A decoder performs corresponding context-adaptive arithmetic decoding.

Entropy encoding and decoding using direct level and run-length/level context-adaptive arithmetic coding/decoding modes

The invention includes several techniques and tools, which can be used in combination or separately. For example, an audio encoder can encode information directly using coding processes that include a windowed overlapped transform, a selective multi-channel transform, scalar quantization and entropy encoding. The audio encoder can also encode information parametrically according to a parametric compression mode that accounts for audibility of distortion according to an auditory model. A corresponding audio decoder can decode first information directly and second information according to the parametric decompression mode.

Multi-channel audio encoding/decoding with parametric compression/decompression and weight factors

An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform.

Complex-transform channel coding with extended-band frequency coding

A communications device is presented for providing bi-directional audio communications between a near-end user and a far-end user via a bidirectional communications channel. The communications device includes an adaptive echo canceller receiving a near-end audio signal and a far-end audio signal and providing an echo-canceled near-end audio signal for transmission to the far-end user via the communications channel. The adaptive echo canceller includes a first bank of analysis filters for filtering the near-end audio signal, a second bank of analysis filters for filtering the far-end audio signal, and a bank of synthesis filters for filtering sub-band echo-canceled signals generated within the adaptive echo canceller. The first and second filter banks have a frequency response optimized to reduce echo residual gain.

Filter bank optimization for acoustic echo cancellation

Described techniques and tools include techniques and tools for mapping digital media data (e.g., audio, video, still images, and/or text, among others) in a given format to a transport or file container format useful for encoding the data on optical disks such as digital video disks (DVDs). A digital media universal elementary stream can be used to map digital media streams (e.g., an audio stream, video stream or an image) into any arbitrary transport or file container, including optical disk formats, and other transports, such as broadcast streams, wireless transmissions, etc. The information to decode any given frame of the digital media in the stream can be carried in each coded frame. A digital media universal elementary stream includes stream components called chunks. An implementation of a digital media universal elementary stream arranges data for a media stream in frames, the frames having one or more chunks.

Digital media universal elementary stream

A spatial element is added to communications, including over telephone conference calls heard through headphones or a stereo speaker setup. Functions are created to modify signals from different callers to create the illusion that the callers are speaking from different parts of the room.

Spatialized audio over headphones

Speech quality estimation technique embodiments are described which generally involve estimating the human speech quality of an audio frame in a single-channel audio signal. A representation of a harmonic component of the frame is synthesized and used to compute a non-harmonic component of the frame. The synthesized harmonic component representation and the non-harmonic component are then used to compute a harmonic to non-harmonic ratio (HnHR). This HnHR is indicative of the quality of a user's speech and is designated as an estimate of the speech quality of the frame. In one implementation, the HnHR is used to establish a minimum speech quality threshold below which the quality of the user's speech is considered unacceptable. Feedback to the user is then provided based on whether the HnHR falls below the threshold.

Harmonicity-based single-channel speech quality estimation

The disclosed architecture employs signal processing techniques to provide audio perception only, or audio perception that matches the visual perception. This also provides spatial audio reproduction for multiparty teleconferencing such that the teleconferencing participants perceive themselves as if they were sitting in the same room. The solution is based on the premise that people perceive sounds as a reconstructed wavefront, and hence, the wavefronts are used to provide the spatial perceptual cues. The differences between the spatial perceptual cues derived from the reconstructed wavefront of sound waves and the ideal wavefront of sound waves form an objective metric for spatial perceptual quality, and provide the means of evaluating the overall system performance. Additionally, compensation filters are employed to improve the spatial perceptual quality of stereophonic systems by optimizing the objective metrics.

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Sammamish, WA, United States of America

Wei-Ge Chen