Abstract: Methods and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. The disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. The distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein a combined cue code (e.g., a combined inter-channel correlation (ICC) code) is generated by combining two or more estimated cue codes, each estimated cue code estimated from a group of two or more channels. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels. Received cue codes include a combined cue code (e.g., a combined ICC code). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein two or more derived cue codes are derived from the combined cue code, and each derived cue code is applied to generate two or more synthesized channels.

Abstract: A method and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. Thus, the disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. Generally, the distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.

Abstract: In a microphone signal, the signal component corresponding to, e.g., echo is suppressed using an echo control scheme that estimates the spectral envelope of the echo signal, without having to estimate the waveform for the echo signal. In one embodiment, the input signal (to be applied to a loudspeaker) and the microphone signal are spectrally decomposed into multiple subbands, where echo suppression processing is independently performed on each subband. The echo control of the present invention can be implemented with substantially reduced (1) computational complexity and (2) phase sensitivity, as compared to traditional acoustic echo cancellation, in which the waveform for the echo signal is estimated.

Abstract: A binaural cue coding scheme in which cue codes are derived from the transmitted audio signal. In one embodiment, an encoder downmixes C input channels to generate E transmitted channels, where C>E>1. A decoder derives cue codes from the transmitted channels and uses those cue codes to synthesize playback channels. For example, in one 5-to-2 BCC embodiment, the encoder downmixes a 5-channel surround signal to generate left and right channels of a stereo signal. The decoder derives stereo cues from the transmitted stereo signal, maps those stereo cues to surround cues, and applies the surround cues to the transmitted stereo channels to generate playback channels of a 5-channel synthesized surround signal.

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E?1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

Abstract: An input audio signal having an input temporal envelope is converted into an output audio signal having an output temporal envelope. The input temporal envelope of the input audio signal is characterized. The input audio signal is processed to generate a processed audio signal, wherein the processing de-correlates the input audio signal. The processed audio signal is adjusted based on the characterized input temporal envelope to generate the output audio signal, wherein the output temporal envelope substantially matches the input temporal envelope.

Abstract: An auditory scene is synthesized from a mono audio signal by modifying, for each critical band, an auditory scene parameter (e.g., an inter-aural level difference (ILD) and/or an inter-aural time difference (ITD)) for each sub-band within the critical band, where the modification is based on an average estimated coherence for the critical band. The coherence-based modification produces auditory scenes having objects whose widths more accurately match the widths of the objects in the original input auditory scene.

Abstract: An apparatus for generating a multi-channel output signal performs a center channel cancellation to obtain improved base channels for reconstructing left-side output channels or right-side output channels. In particular, the apparatus includes a cancellation channel calculator for calculating a cancellation channel using information related to the original center channel available at the decoder. The device furthermore includes a combiner for combining a transmission channel with the cancellation channel. Finally, the apparatus includes a reconstructor for generating the multi-channel output signal. Due to the center channel cancellation, the channel reconstructor not only uses a different base channel for reconstructing the center channel but also uses base channels different from the transmission channels for reconstructing left and right output channels which have a reduced or even completely cancelled influence of the original center channel.

Abstract: For a multi-channel audio signal, parametric coding is applied to different subsets of audio input channels for different frequency regions. For example, for a 5.1 surround sound signal having five regular channels and one low-frequency (LFE) channel, binaural cue coding (BCC) can be applied to all six audio channels for sub-bands at or below a specified cut-off frequency, but to only five audio channels (excluding the LFE channel) for sub-bands above the cut-off frequency. Such frequency-based coding of channels can reduce the encoding and decoding processing loads and/or size of the encoded audio bitstream relative to parametric coding techniques that are applied to all input channels over the entire frequency range.

Abstract: A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

Abstract: The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front/back coherence.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: A method and apparatus are disclosed for representing the masked threshold in a perceptual audio coder, using line spectral frequencies (LSF) or another representation for linear prediction (LP) coefficients. The present invention calculates LP coefficients for the masked threshold using known LPC analysis techniques. In one embodiment, the masked thresholds are optionally transformed to a non-linear frequency scale suitable for auditory properties. The LP coefficients are converted to line spectral frequencies or a similar representation in which they can be quantized for transmission. In one implementation, the masked threshold is transmitted only if the masked threshold is significantly different from the previous masked threshold. In between each transmitted masked threshold, the masked threshold is approximated using interpolation schemes.

Abstract: In a microphone signal, the signal component corresponding to, e.g., echo is suppressed using an echo control scheme that estimates the spectral envelope of the echo signal, without having to estimate the waveform for the echo signal. In one embodiment, the input signal (to be applied to a loudspeaker) and the microphone signal are spectrally decomposed into multiple subbands, where echo suppression processing is independently performed on each subband. The echo control of the present invention can be implemented with substantially reduced (1) computational complexity and (2) phase sensitivity, as compared to traditional acoustic echo cancellation, in which the waveform for the echo signal is estimated.

Abstract: A perceptual audio coder is disclosed for encoding audio signals, such as speech or music, with different spectral and temporal resolutions for the redundancy reduction and irrelevancy reduction using cascaded filterbanks. The disclosed perceptual audio coder includes a first analysis filterbank for performing irrelevancy reduction in accordance with a psychoacoustic model and a second analysis filterbank for performing redundancy reduction. The spectral/temporal resolution of the first filterbank can be optimized for irrelevancy reduction and the spectral/temporal resolution of the second filterbank can be optimized for maximum redundancy reduction. The disclosed perceptual audio coder also includes a scaling block between the cascaded filterbank that scales the spectral coefficients, based on the employed perceptual model.

Abstract: An auditory scene is synthesized from a mono audio signal by modifying, for each critical band, an auditory scene parameter (e.g., an inter-aural level difference (ILD) and/or an inter-aural time difference (ITD)) for each sub-band within the critical band, where the modification is based on an average estimated coherence for the critical band. The coherence-based modification produces auditory scenes having objects whose widths more accurately match the widths of the objects in the original input auditory scene.

Abstract: Noise-like signal components are detected within arbitrary regions of the time-frequency plane. Various transforms are applied to G time domain samples with different spectral/temporal resolutions. The flatness of the time domain samples and the frequency samples for each transform are compared. If the computed flatness measures are about the same, the samples are assumed to be noisy. Noise-like signal components can be detected using a general filterbank within a limited time interval and frequency range by decomposing the signal into N subbands. To each group of G subband samples in time {tk}, a linear orthogonal transform is applied to obtain the frequency domain samples {fi}. The flatness of the time domain samples is compared to the flatness of the frequency domain samples {fi}. A filterbank with uniform frequency-tiling can be used to detect noise-like signal components.

Abstract: A method and apparatus are disclosed for controlling a buffer in a communication system, such as a digital audio broadcasting (DAB) communication system. A more consistent perceptual quality over time provides for a more pleasing auditory experience to a listener. Thus, the disclosed bit allocation process determines, for each frame, a distortion d[k] at which the frame is to be encoded. Generally, the distortion d[k] is determined to minimize (i) the probability for a buffer overflow, and (ii) the variation of perceived distortion over time. A buffer level is controlled by partitioning a signal into a sequence of successive frames; estimating a distortion rate for a number of frames; and selecting a distortion such that the variance of the buffer level is bounded by a specified value.

Abstract: Perceptual coding of spatial cues (PCSC) is used to convert two or more input audio signals into a combined audio signal that is embedded with two or more sets of one or more auditory scene parameters, where each set of auditory scene parameters (e.g., one or more spatial cues such as an inter-ear level difference (ILD), inter-ear time difference (ITD), and/or head-related transfer function (HRTF)) corresponds to a different frequency band in the combined audio signal. A PCSC-based receiver is able to extract the auditory scene parameters and apply them to the corresponding frequency bands of the combined audio signal to synthesize an auditory scene. The technique used to embed the auditory scene parameters into the combined signal enables a legacy receiver that is unaware of the embedded auditory scene parameters to play back the combined audio signal in a conventional manner, thereby providing backwards compatibility.