Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Techniques are described for managing synchronized jitter buffers for streaming data (e.g., for real-time audio and/or video communications). A separate jitter buffer can be maintained for each codec. For example, as data is received in network packets, the data is added to the jitter buffer corresponding to the codec that is associated with the received data. When data needs to be read, the same amount of data is read from each of the jitter buffers. In other words, at each instance where data needs to be obtained (e.g., for decoding and playback), the same amount of data is obtained from each of the jitter buffers. In addition, the multiple jitter buffers use the same playout timestamp that is synchronized across the multiple of jitter buffers.

Abstract: Techniques are described for performing forward and backward extrapolation of data to compensate for data that has been lost due to network packet loss. The forward and backward extrapolation can be used to perform packet loss concealment. For example, when network packet loss is detected, network packets before and after the lost data can be identified. Forward and backward extrapolation can then be applied to cover the period of lost data. For example, the network packets before the period of lost data can be used to perform forward extrapolation to cover a first portion of the period of lost data. The network packets after the period of lost data can be used to perform backward extrapolation to cover a remaining portion of the period of lost data. The period of lost data can be reconstructed based at least in part on the extrapolation.

Abstract: Techniques are described for performing conditional forward error correction (FEC) of network data. The techniques and solutions can be applied to suppress the transmission of redundant forward error correction information for data (e.g., frames of audio and/or video data) that can be effectively recovered at the receiving device (e.g., at the decoder). For example, a first computing device that is encoding and transmitting data (e.g., encoded audio data) to a second computing device can determine whether portions of data can be predicted (e.g., to a certain quality measure) at the second computing device. If the portions of data can be predicted, then the first computing device can skip sending redundant copies of the portions of data (e.g., can skip sending forward error correction information) in current network packets.

Abstract: Techniques are described for managing synchronized jitter buffers for streaming data (e.g., for real-time audio and/or video communications). A separate jitter buffer can be maintained for each codec. For example, as data is received in network packets, the data is added to the jitter buffer corresponding to the codec that is associated with the received data. When data needs to be read, the same amount of data is read from each of the jitter buffers. In other words, at each instance where data needs to be obtained (e.g., for decoding and playback), the same amount of data is obtained from each of the jitter buffers. In addition, the multiple jitter buffers use the same playout timestamp that is synchronized across the multiple of jitter buffers.

Abstract: Techniques are described for determining corrected timestamps for streaming data that is encoded using frames with a variable frame size. The streaming data is encoded into frames and transmitted in network packets in which the network packets or frames are associated with timestamps incremented in fixed steps. When a network packet is received after a lost packet, a corrected timestamp range can be calculated for the received packet based at least in part on the received timestamp value and attributes of the received network packet along with buffering characteristics.

Abstract: Techniques are described for performing conditional forward error correction (FEC) of network data. The techniques and solutions can be applied to suppress the transmission of redundant forward error correction information for data (e.g., frames of audio and/or video data) that can be effectively recovered at the receiving device (e.g., at the decoder). For example, a first computing device that is encoding and transmitting data (e.g., encoded audio data) to a second computing device can determine whether portions of data can be predicted (e.g., to a certain quality measure) at the second computing device. If the portions of data can be predicted, then the first computing device can skip sending redundant copies of the portions of data (e.g., can skip sending forward error correction information) in current network packets.

Abstract: Techniques are described for performing forward and backward extrapolation of data to compensate for data that has been lost due to network packet loss. The forward and backward extrapolation can be used to perform packet loss concealment. For example, when network packet loss is detected, network packets before and after the lost data can be identified. Forward and backward extrapolation can then be applied to cover the period of lost data. For example, the network packets before the period of lost data can be used to perform forward extrapolation to cover a first portion of the period of lost data. The network packets after the period of lost data can be used to perform backward extrapolation to cover a remaining portion of the period of lost data. The period of lost data can be reconstructed based at least in part on the extrapolation.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Innovations in phase quantization during speech encoding and phase reconstruction during speech decoding are described. For example, to encode a set of phase values, a speech encoder omits higher-frequency phase values and/or represents at least some of the phase values as a weighted sum of basis functions. Or, as another example, to decode a set of phase values, a speech decoder reconstructs at least some of the phase values using a weighted sum of basis functions and/or reconstructs lower-frequency phase values then uses at least some of the lower-frequency phase values to synthesize higher-frequency phase values. In many cases, the innovations improve the performance of a speech codec in low bitrate scenarios, even when encoded data is delivered over a network that suffers from insufficient bandwidth or transmission quality problems.

Abstract: Techniques are described for determining corrected timestamps for streaming data that is encoded using frames with a variable frame size. The streaming data is encoded into frames and transmitted in network packets in which the network packets or frames are associated with timestamps incremented in fixed steps. When a network packet is received after a lost packet, a corrected timestamp range can be calculated for the received packet based at least in part on the received timestamp value and attributes of the received network packet along with buffering characteristics.

Abstract: A first user terminal, host terminal, method and program. The first terminal comprises: a transceiver for communicating with a plurality of other user terminals over a communication network; and communications processing apparatus, coupled to the transceiver, and arranged to participate in a call with a selected number of the other user terminals via the transceiver and communication network, the call including transmission of a voice signal from the first user terminal. The communications processing apparatus is operable in a mode whereby it temporarily discontinues transmission of the voice signal in response to detecting less than a predetermined level of activity on said voice signal, and the communications processing apparatus is further configured to selectively enable that mode in dependence on the selected number of other user terminals in the call.

Abstract: A far end signal is received at a device, a marker signal is inserted into the far end signal and the far end signal with the marker signal is played on a speaker. A near end signal is received via a microphone and the marker signal is detected in said received near end signal. The detected marker signal is used to determine a delay that is then used to cancel at least some of an echo in the near end signal. The marker may be ultrasonic. The echo canceller and other processing may run at a lower sampling frequency than the marker detection.

Abstract: Processing of a signal received at a node in a network is described in which effects on the signal caused by applying an action to a first part of the signal are quantified based on characteristics of the first part of the signal and effects on the signal caused by not applying the action to the first part of the signal are quantified based on characteristics of a second, subsequent part of the signal. The action may then be selectively applied either to the first part of the signal or to the second part of the signal based upon the quantifications. In some embodiments, the action is applied to a portion of the signal for which the effects on at least one measure of the signal quality are less detrimental.