Enhanced Variable Rate Codec

About: Enhanced Variable Rate Codec is a research topic. Over the lifetime, 357 publications have been published within this topic receiving 4842 citations. The topic is also known as: EVRC.

Method and system for terminal codec setup of multimedia ring back tone service

Abstract: Disclosed is a method for terminal codec setup for a multimedia Ring Back Tone (RBT) service, which allows RBT sound sources previously set in a sound source-providing server by a called subscriber to be reproduced to an originating terminal by means of a Home Location Register (HLR) and the sound source-providing server for storing the RBT sound sources when a calling subscriber telephones the called subscriber, the HLR storing profile information including whether the subscriber has joined the RBT service. The method includes the steps of : (a) receiving a first codec setup message including information (multimedia codec information) regarding a multimedia codec from the called subscriber, after transmitting an ISDN User Part (ISUP) call connection request message including the multimedia codec information to the called subscriber; (b) when the first codec setup message is received, transmitting a second codec setup message for requesting setup of the multimedia codec to a call-side Base Transceiver Station (BTS) , thereby controlling a call-side vocoder located in a call-side Base Station Controller (BSC) to set the multimedia codec; (c) when the first codec setup message is received, transmitting a third codec setup message for requesting setup of the multimedia codec to the originating terminal, thereby controlling the originating terminal to set the multimedia codec; and (d) receiving an RBT sound source selected using the multimedia codec information from the sound source- providing server and transmitting the RBT sound source to the originating terminal.

A 16-kbit/s wideband speech codec scalable with g.729.

Abstract: A wideband speech scalable codec is proposed for improving the flexibility in telecommunication networks. This coder is scalable with G.729 (ITU 8-kbit/s standard). Its decoder can process the incoming bitstream at three bit rates (8, 12, and 16 kbit/s) and provide a choice of speech types (wideband and telephone-band). The codec has a split-band structure, where both bands are coded by analysis-by-synthesis techniques. This paper proposes two types of scalable codec: a separate one and a composite one. It also proposes a new method (an additional adaptive codebook) for predicting pitch, while maintaining scalability with the G.729 codec. Subjective testing for wideband speech showed that the quality of the proposed codec at 16-kbit/s is equivalent to that of the 64-kbit/s G.722, and at 12-kbit/s is better than that of the 48-kbit/s G.722. Testing has further demonstrated that the 8-kbit/s coder provides high quality for telephone-band speech.

Multichannel coding of applause signals

Abstract: We develop a parametric multichannel audio codec dedicated to coding signals consisting of a dense series of transient-type events. These signals of which applause is a typical example are known to be problematic for such audio codecs. The codec design is based on preservation of both timbre and transient-type event density. It combines a very low complexity and a low parameter bit rate (0.2 kbps). In a formal listening test, we compared the proposed codec to the recently standardised MPEG Surround multichannel codec, with an associated parameter bit rate of 9 kbps. We found the new codec to have a significantly higher audio quality than the MPEG Surround codec for the two multichannel applause signals under test. Though this seems promising, the technique presented is not fully mature, for example, because issues related to integration of the proposed codec in the MPEG Surround codec were not addressed.

A high-fidelity speech and audio codec with low delay and low complexity

Abstract: This paper presents a high-fidelity speech and audio codec operating at a sampling rate of 32 kHz and a bit rate of 64 kbit/s. Designed primarily for real-time speech communication systems with high port densities, this MDCT-based transform codec has a very low coding delay (8 ms frame size) and low codec complexity (less than 10 MIPS on a 16-bit fixed-point DSP). The codec achieves essentially transparent quality for speech, and very close to transparent quality for music. A novel frame erasure concealment algorithm makes this codec robust to frame erasures for both speech and music. Another novel feature allows the decoder to decode the bit stream directly into a 16 kHz or 8 kHz sampled signal, without the need to decode a 32 kHz signal first and then down-sample it to the target sampling rate. Other novel features include some speed-memory trade-off techniques to reduce the computational complexity.

Standardization of the selectable mode vocoder

Abstract: The migration from the first generation of cellular telephony to the second has included a transition from an analog speech channel to a digital channel that employs digital speech codecs. As the deployment of these second-generation systems matures, system capacity concerns have increased the pressure for a more efficient encoding of speech. In addition, market pressures have contributed to the contradictory requirement of improved voice quality provided by these wireless systems. This paper presents the motivation for, and the execution of, a program of standardizing a new variable-rate speech codec for the cdmaOne/cdma2000 wireless system. Several codecs have previously been standardized. This new codec, known as the selectable mode vocoder or SMV, offers a significant improvement in voice quality over that of existing codec standards as well as the flexibility of allowing the system operator to make tradeoffs between voice quality and system capacity.