Topic
Adaptive Multi-Rate audio codec
About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.
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07 Apr 1986
TL;DR: A 7 kHz-band speech CODEC using an advanced digital signal processor VLSI, "DSSP-1", is developed and arithmetic word-length, arithmetic operation speed, and memory capacity required for this C ODEC is evaluated.
Abstract: A 7 kHz-band speech CODEC using an advanced digital signal processor is developed. The coding algorithm for this CODEC requires greater processing power due to its complexity and high sampling rate. A high-performance digital signal processor VLSI, "DSSP-1", is applied to this CODEC. Arithmetic word-length, arithmetic operation speed, and memory capacity required for this CODEC is evaluated. DSSP-1 can satisfy these requirements; this CODEC has been built with two DSSP-1 chips and several memory chips.
07 Apr 2023
TL;DR: In this article , a psychoacoustic model was integrated with the existing structure that contains a convolutional encoder, decoder, and a residual vector quantizer to enhance the quality of neural audio codecs.
Abstract: Neural audio codecs are the most recent development in the field of audio compression. Traditional audio codecs rely on fixed signal processing pipelines and require domain-specific expertise to produce high-quality audio at low to high bit rates. However, the performance of conventional audio codecs usually degrades at low bit rates. Neural audio codecs perform enhancement and compression with no added latency. This paper further enhances the quality of neural audio codecs by integrating a psychoacoustic model with the existing structure that contains a convolutional encoder, decoder, and a residual vector quantizer. It used a combination of reconstruction and adversarial loss to train the model to generate high-quality audio content. Audio quality measures like PEAQ and MUSHRA are conducted to illustrate that the proposed model performs better than the existing model of neural audio codec.
TL;DR: This article examines an original process to code signals from two different sources mixed together, and coded within a single coding process (DCT-based JPEG codec) in order to generate a single data stream at its output, instead of two as in the usual multimedia applications.
Abstract: This article examines an original process to code signals from two different sources (data e.g. audio- and video) mixed together, and coded within a single coding process (DCT-based JPEG codec), in order to generate a single data stream at its output, instead of two as in the usual multimedia applications.
11 Jun 1991
TL;DR: A comparative evaluation between the full-rate and the half-rate source codecs is presented: the speech quality is almost the same on a wide range of experiments and different speakers.
Abstract: The authors describe a real-time implementation of a voiceband codec suitable for the GSM Half-Size Digital Mobile Radio (DMR) system at 900 MHz. This codec exploits the CELP (code excited linear prediction) algorithm running at 6.3 kbit/s in connection with a channel coding scheme, adding 5.1 kbit/s of redundancy to reach the final rate of 11.4 kbit/s for the traffic channel. A comparative evaluation between the full-rate and the half-rate source codecs is presented: the speech quality is almost the same on a wide range of experiments and different speakers. >
04 Sep 2000
TL;DR: This paper discusses the design and implementation of a low bit rate codec that has a low delay and greater compression, even though the techniques used are simpler and is suitable for applications like video conferencing, mobile communications etc.
Abstract: This paper discusses the design and implementation of a low bit rate codec along with its performance at different bit rates. The International Telecommunications Union's G.728 CELP speech coder is specifically designed for low coding delay and toll quality speech at a rate of 16kbps. Here, we present the design of a CELP(Code Excited Linear Prediction) algorithm similar to the above coder, but can compress speech up to rates like 6.4kbps(as in the G.723.1 standard). This codec shares the advantages of both the above standards; it has a low delay (up to 1.25ms) and greater compression, even though the techniques used are simpler. This advantage makes this coder suitable for applications like video conferencing, mobile communications etc.