Networked music performance (NMP) is a potential game changer among Internet applications, as it aims at revolutionizing the traditional concept of musical interaction by enabling remote musicians to interact and perform together through a telecommunication network. Ensuring realistic performance conditions, however, constitutes a significant engineering challenge due to the extremely strict requirements in terms of network delay and audio quality, which are needed to maintain a stable tempo, a satisfying synchronicity between performers and, more generally, a high-quality interaction experience. In this paper, we offer a review of the psycho-perceptual studies conducted in the past decade, aimed at identifying latency tolerance thresholds for synchronous real-time musical performance. We also provide an overview of hardware/software enabling technologies for NMP, with a particular emphasis on system architecture paradigms, networking configurations, and applications to real use cases.

An Overview on Networked Music Performance Technologies

The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Audio encoder and decoder

An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain Furthermore, the audio encoder has a second coding branch comprising a second time/frequency converter Additionally, a signal analyzer for analyzing the audio signal is provided The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch On the other hand, the signal analyzer determines a time/frequency resolution to be applied by the converters when generating the encoded signals An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time/frequency converter and used by the second time/frequency converter

Multi-resolution switched audio encoding/decoding scheme

An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Audio encoder, audio decoder and audio processor having a dynamically variable warping characteristic

With increasing quality requirements for multimedia communications, audio codecs must maintain both high quality and low delay. Typically, audio codecs offer either low delay or high quality, but rarely both. We propose a codec that simultaneously addresses both these requirements, with a delay of only 8.7 ms at 44.1 kHz. It uses gain-shape algebraic vector quantization in the frequency domain with time-domain pitch prediction. We demonstrate that the proposed codec operating at 48 kb/s and 64 kb/s out-performs both G.722.1C and MP3 and has quality comparable to AAC-LD, despite having less than one fourth of the algorithmic delay of these codecs.

/pdf/a-high-quality-speech-and-audio-codec-with-less-than-10-ms-2z9yw1jf5z.pdf

A High-Quality Speech and Audio Codec With Less Than 10-ms Delay

An apparatus for encoding comprises a first domain Converter (510), a switchable bypass (50), a second domain Converter (410), a first processor (420) and a second processor (520) to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain Converter allow the generation of a decoded audio signal with high quality and low bit rate.

Audio encoding/decoding scheme having a switchable bypass

An audio encoder for encoding an audio signal includes an impulse extractor (10) for extracting an impulse-like portion from the audio signal. This impulse-like portion is encoded and forwarded to an output interface (22). Furthermore, the audio encoder includes a signal encoder (16) which encodes a residual signal derived from the original audio signal so that the impulse-like portion is reduced or eliminated in the residual audio signal. The output interface (22) forwards both, the encoded signals, i.e., the encoded impulse signal (12) and the encoded residual signal (20) for transmission or storage. On the decoder-side, both signal portions are separately decoded and then combined to obtain a decoded audio signal.

Audio encoder for encoding an audio signal having an impulse-like portion and stationary portion, encoding methods, decoder, decoding method; and encoded audio signal

A very coarse quantization exceeding the measure determined by the masking threshold without or only very little quality losses is enabled by quantizing not immediately the prefiltered signal, but a prediction error obtained by forward-adaptive prediction of the prefiltered signal. Due to the forward adaptivity, the quantizing error has no negative effect on the prediction on the decoder side.

Information signal encoding

The goal of the presented schemes is to obtain an ultra low delay audio coder with a good performance even at low bit rates (around 64 kb/s). The problem to be solved is to gain sufficient frequency resolution at low frequencies for precise low frequency psycho-acoustics and quantization noise shaping, because the ear has a higher frequency resolution at lower frequencies. Our approach is to use warped linear noise shaping pre- and post-filters, and a short DFT for the psycho-acoustic model (length 256), but with frequency warping. We compare four different psychoacoustic versions: DFT with no warping; DFF without warping using warped pre- and post-filters; warping with the so-called NDFT (WDFT); DFT with all-pass delay chain pre-processing. Listening tests show that the best performance is obtained using WDFT.

Frequency warping in low delay audio coding

Reduced Bit Rate Ultra Low Delay Audio Coding

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