Microsoft Technology Licensing, LLC

Microsoft Corporation

Chao He

Wei-Ge Chen

Qin Li

Sanjeev Mehrotra

Kazuhito Koishida

Vinod Prakash

Naveen Thumpudi

Wei-ge Chen

Jack W Stokes

Anton W Krantz

Michael Lewis Seltzer

Arungunram Chandrasekaran Surendran

Mu Han

Wei Zhong

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

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

A voice communication end device performs quality checks to determine whether acoustic echo cancellation would be ineffective, such as due to noise or clock drift or discontinuities between incoming and outgoing voice channels. In the case where echo cancellation would prove ineffective, the device falls back on a tri-state voice switching operation that includes a bi-direction state in which both channels are on in full duplex operation, which provides a smoother transition switching between active channels. The tri-state voice switching supports both voluntary transitions where the active user voluntarily stops to yield the active channel, and forced transitions where the active user is forcedly interrupted by the other user speaking more loudly.

Voice switching for voice communication on computers

Architecture that mitigates echo in voice communications using echo detection and adaptive management of attenuation by a non-linear processor (NLP). Suppression values provided by the NLP are determined based on echo detection and retained on a case-by-case basis to automatically increase or decrease the attenuation as needed. Feedback is incorporated that where the controls for the NLP attenuation is given by the amount of echo that remains, and this in turn affects the amount of echo that remains.

Mitigation of echo in voice communication using echo detection and adaptive non-linear processor

A mixed lossless audio compression has application to a unified lossy and lossless audio compression scheme that combines lossy and lossless audio compression within a same audio signal. The mixed lossless compression codes a transition frame between lossy and lossless coding frames to produce seamless transitions. The mixed lossless coding performs a lapped transform and inverse lapped transform to produce an appropriately windowed and folded pseudo-time domain frame, which can then be losslessly coded. The mixed lossless coding also can be applied for frames that exhibit poor lossy compression performance.

Mixed lossless audio compression

Presence of echo in a conversation is determined through a search performed in the sub-band domain using independent short length adaptive filters across a range of sub-band and maximum expected echo delay values. The adaptive filters attempt to predict and cancel the echo in the microphone signal based on the content in the speaker signal. If substantial cancellation is achieved in any of the sub-band filters, echo presence in the microphone signal is determined.

Echo presence determination in voice conversations

A voice communication end device provides echo reduction when operating in a full duplex mode using acoustic echo cancellation, which includes periodic audio queue adjustments to account for drift. The end device performs a quality assessment by calculating consistency statistics for the queue adjustments to determine whether acoustic echo cancellation would be ineffective, and if so falls back to a half duplex mode using voice switching.

Timestamp quality assessment for assuring acoustic echo canceller operability

The quality and robustness of audio echo cancellation is enhanced by selectively applying glitch recovery processes based on a quality measurement of the relative offset between capture and render audio streams. For example, large and small glitch detection is enabled for low relative offset variance; large glitch detection is enabled in a medium range of relative offset variance; and neither enabled at high variance. Further, a fast glitch recovery process suspends updating the adaptive filter coefficients of the audio echo cancellation while buffers are re-aligned to recover from the glitch, so as to avoid resetting the adaptive filter. When clock drift exists between capture and render audio streams, a multi-step compensation method is applied to improve AEC output quality in case the drifting rate is low; and a resampler is used to compensate the drift in case the drifting rate is high. An anti-clipping process detects clipping of the signals, and also suspends adaptive filter updating during clipping.

Selective glitch detection, clock drift compensation, and anti-clipping in audio echo cancellation

A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

Method and apparatus for reducing timestamp noise in audio echo cancellation

An audio signal is received that might include keyboard noise and speech. The audio signal is digitized and transformed from a time domain to a frequency domain. The transformed audio is analyzed to determine whether there is likelihood that keystroke noise is present. If it is determined there is high likelihood that the audio signal contains keystroke noise, a determination is made as to whether a keyboard event occurred around the time of the likely keystroke noise. If it is determined that a keyboard event occurred around the time of the likely keystroke noise, a determination is made as to whether speech is present in the audio signal around the time of the likely keystroke noise. If no speech is present, the keystroke noise is suppressed in the audio signal. If speech is detected in the audio signal or if the keystroke noise abates, the suppression gain is removed from the audio signal.

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Redmond, WA, United States of America

Chao He