An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder. A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.

Audio encoding and 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

A port adapter for connecting zero or more network interfaces to a host system having a SPI-4 bus is disclosed. The port adapter comprises zero or more network interfaces; a SPI-4 bus coupled to a host system to provide a communication channel between the host and the network interfaces; a control bus coupled to the host system for controlling and monitoring the port adapter; and interface logic that interfaces the SPI-4 bus and the control bus to the network interfaces. Methods are provided for selecting and using one of a small plurality of different packet formats for various networking technologies, so that the port adapter can hide details of the technology that it handles from the host system, and for operating the host system's SPI-4 bus at one of several speeds based on bandwidth requirements of the port adapter.

Port adapter for high-bandwidth bus

An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Multi-channel audio encoding and decoding

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Parametric joint-coding of audio sources

A multi-channel linear predictive analysis-by-synthesis signal encoding method detects (S26, S27) inter-channel correlation and select one of several possible encoding modes (S24, S29, S30) based on the detected correlation.

Multi-channel signal encoding and decoding

A speech quality estimation technique that employs an arbitrary, speech quality estimation algorithm. The speech quality estimation technique analyzes a reference speech signal and a test speech signal, and based on this analysis, identifies the level of continuous delay variation, if any, and the location of and size of any intermittent delay variations along the test signal. The reference speech signal and/or the test speech signal are adjusted to account for continuous delay variation and intermittent delay variations, such that the reference speech signal and the test signal are similarly scaled with respect to the time domain. The reference speech signal and the test speech signal are then compared for the purpose of generating a speech quality estimation. The resulting speech quality estimation is then adjusted based on the level of continuous delay variation and any intermittent delay variations.

Handling variable delay in objective speech quality assessment

Measuring the quality of speech signals in a telecommunications system is more effectively accomplished by first estimating and eliminating the effects of linear filtering that are associated with the telecommunications system. In doing so, a frequency domain representation of a test speech signal and a reference speech signal are generated. The frequency domain representations may take the form of bark-time frame matrices. Each of a plurality of frequency weights are then estimated based on the frequency domain representations of the test and reference speech signals, where each of the plurality of frequency weights corresponds with a different frequency band, and where the frequency weights reflect the linear filter characteristics of the telecommunications systems. The frequency domain representation of the reference signal is then adjusted by applying each of the frequency weights to a corresponding component of the frequency domain representation of the reference signal. The frequency representation of the test signal and the adjusted frequency domain representation of the reference signal are compared, and speech quality is measured based on the comparison.

Compensation for linear filtering using frequency weighting factors

A multi-part fixed codebook includes both individual fixed codebooks (FC1, FC2) for each channel and a shared fixed codebook (FCS). Although the shared fixed codebook (FCS) is common to all channels, the channels are associated with individual lags (D1, D2). Furthermore, the individual fixed codebooks (FC1, FC2) are associated with individual gains (gF1, gF2), and the individual lags (D1, D2) are also associated with individual gains (gFS1, gFS2). The excitation from each individual fixed codebook (FS1, FS2) is added to the corresponding excitation (a shared codebook vector, but individual lags and gains for each channel) from the shared fixed codebook (FCS).

Multi-channel signal coding

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Multi-channel signal encoding and decoding

Handling variable delay in objective speech quality assessment

Compensation for linear filtering using frequency weighting factors

Multi-channel signal coding