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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30 Jul 2000
TL;DR: A low bit rate, standalone and portable video communication system for wire and wireless applications, deployed three general purpose DSP for the real time implementation of video coding, audio coding, channel coding and system control.
Abstract: We present a low bit rate, standalone and portable video communication system for wire and wireless applications. The video codec is based on H.263, and the audio codec is based on G.723.1. When the application is for wireless the transmission bandwidth is limited to 25 kHz. In order to have reasonably good video and audio quality, the source bit rate is set to 38 kbps. This includes video, audio and synchronization. The bit stream is embedded with 26 kbps for the error protection, resulting in a total of 64 kbps. A 16-QAM is used for the modulation in order to meet the bandwidth requirement. Besides, the RF module is developed for the transmission on the UHF band. An interface for a normal modem is also implemented in the system to make the wired application available. The whole system deployed three general purpose DSP for the real time implementation of video coding, audio coding, channel coding and system control.
Patent•
05 Nov 2014
TL;DR: In this article, an audio coding method was proposed to improve the linear prediction efficiency of a current audio frame by matching the audio coding methods matching the linear predictions of the current frame.
Abstract: An audio coding method, comprising: estimating the referential linear prediction efficiency of a current audio frame (101); determining an audio coding method matching the referential linear prediction efficiency of the current audio frame (102); and conducting audio coding on the current audio frame according to the audio coding method matching the referential linear prediction efficiency of the current audio frame (103) The technical solution facilitates decreasing audio coding overhead
22 Oct 1998
TL;DR: Two schemes for speech processing are proposed, which have variable transmission delay and bit rate respectively, to reach at the goals: small assembly delay, high transmission efficiency and toll speech quality.
Abstract: There are some new demands on speech processing when voice communication is carried out over ATM networks. This paper discusses some problems of speech coding and transmission on ATM networks. We propose two schemes for speech processing, which have variable transmission delay and bit rate respectively, to reach at the goals: small assembly delay, high transmission efficiency and toll speech quality. We transmit different speech cells in different time intervals and define a new "distribution entropy" to classify different input signals. We then design a new speech codec in the second scheme whose bit rate is perceptual importance based. It has better speech quality than that of the G.727 system.
TL;DR: In this article , the authors proposed a novel method for low-rate wideband speech coding utilizing a standard narrowband codec, which can achieve subjective quality comparable to the speeches coded by wideband codecs at higher bitrates in a subjective MUSHRA test.
Abstract: Decimation of a discrete-time signal below the Nyquist rate without applying an appropriate lowpass filter results in a distortion called aliasing. If wideband speech sampled at 16 kHz is decimated by 2 to result in a signal sampled at 8 kHz with aliasing, the decimated signal would be the summation of two speech-like signals, which are the narrowband speech covering 0-4 kHz and the spectrally flipped aliasing component coming from 8-4 kHz. Recently, the performance of speech separation has been remarkably improved with deep learning-based approaches, implying that the narrowband and aliasing components may be able to be separated. In this letter, we propose a novel method for low-rate wideband speech coding utilizing a standard narrowband codec. Instead of coding wideband speech using a wideband codec with a limited bitrate, we propose to decimate the input wideband speech incurring aliasing, and then encode it with a narrowband codec by allocating all the allowed bitrate to 0-4 kHz. After decoding the encoded bitstream, we apply a speech separation technique to obtain the narrowband and aliasing signals, which are then used to reconstruct the wideband speech by expansion, low/highpass filtering, and summation. Experimental results showed that the proposed method could achieve subjective quality comparable to the speeches coded by wideband codecs at higher bitrates in a subjective MUSHRA test.
09 Jul 2014
TL;DR: Investigation of novel packet loss protection schemes to compress mixtures of speech sources for interactive real-time audio services such as spatial teleconferencing indicates the proposed scheme maintains the perceptual quality of the speech sources across a wide variety of packet loss conditions.
Abstract: This paper investigates the application of novel packet loss protection schemes to compress mixtures of speech sources for interactive real-time audio services such as spatial teleconferencing. Hybrid Forward Error Correction (FEC) and Multiple Description Coding (MDC) packet loss protection techniques are applied to the outputs of a psychoacoustic-based Analysis-By-Synthesis (PABS) coder designed for such applications. The protection approaches split the coder outputs into two descriptions that are separately protected using the hybrid FEC-MDC techniques. Perceptual Evaluation of Speech Quality (PESQ) measurements compare the performance of different protection schemes for a range of typical packet loss conditions. Results indicate the proposed scheme maintains the perceptual quality of the speech sources across a wide variety of packet loss conditions.