Juin-Hwey Chen

The Broadvoice Speech Coding Algorithm

This paper describes the BroadVoice® speech coding algorithm, which has been standardized as PacketCable™, SCTE®, and ANSI standards for VoIP cable telephony. The BroadVoice family of codecs includes a 16 kb/s BroadVoice 16 narrowband codec and a 32 kb/s BroadVoice32 wideband codec. BroadVoice is based on two-stage noise feedback coding with vector quantization of excitation and is optimized for low delay, low complexity, and high quality. It has an algorithmic delay of merely 5 ms and a complexity of only 12 and 17 MIPS for BroadVoice16 and BroadVoice32, respectively. The speech quality of BroadVoice16 is better than that of G.728, G.729, and 32 kb/s G.726. The speech quality of BroadVoice32 is better than that of 64 kb/s G.722.

Packet loss concealment based on extrapolation of speech waveform

A class of packet loss concealment algorithms for speech coding is presented. It generates the replacement waveform for the lost frame by direct extrapolation of the past speech waveform, with or without look-ahead. The ITU-T G.722 Appendix III standard is based on it. When a future frame is unavailable (without look-ahead), the PLC algorithm gives significantly better speech quality than G.711 Appendix I - by about 0.2 PESQ for high packet loss rates. When a future frame is available (with look-ahead), the PLC algorithm uses the decoded speech waveform in the future frame to guide the pitch contour of waveform extrapolation during the lost frame such that the extrapolated waveform is phase-aligned with the decoded waveform after the packet loss. This technique further improved PESQ by another 0.2 for high packet loss rates.

A Candidate for the ITU-T G.722 Packet Loss Concealment Standard

This paper presents a candidate for the ITU-T G.722 packet loss concealment standard. The algorithm is based on waveform extrapolation in the speech domain. The strong backward adaptive nature of G.722 makes the state update during lost frames a challenge. The paper presents methods to update the G.722 subband decoder state memory during packet loss. Furthermore, novel techniques to facilitate smooth transition after packet loss are described. Formal subjective test results indicate that the algorithm far exceeds all requirements in the ITU-T terms of reference. Most techniques are applicable to G.726 as well, and some are easily extendable to other speech coders with memory.

Novel Codec Structures for Noise Feedback Coding of Speech

This paper presents several novel codec structures for noise feedback coding (NFC) incorporating both long-term and short-term noise spectral shaping, as well as long-term and short-term prediction. In addition, the paper generalizes the conventional scalar-quantization-based NFC to vector-quantization-based NFC, and it lays the foundation for the associated efficient VQ codebook search and closed-loop VQ codebook design. BroadVoicereg16, a PacketCable 1.5 mandatory narrowband speech codec standardized by CableLabsreg for Voice over Cable in North America, is based on one of such novel NFC codec structures

Packet Loss Concealment for Predictive Speech Coding Based on Extrapolation of Speech Waveform

Most packet loss concealment (PLC) algorithms for predictive speech coders are based on extrapolating the excitation signal. This paper presents an alternative algorithm that extrapolates the speech waveform directly. The algorithm avoids the additional delay associated with the overlap-add operation in the ITU-T G.711 Appendix I by utilizing the ;ringing; of the synthesis filter. It also performs special operations to update the filter states to prepare the predictive decoder for the next good frame. This algorithm achieves robust PLC performance when applied to the BroadVoicereg standard coders. It can also be applied to other predictive speech coders.

BroadVoice® 16: A PacketCable Speech Coding Standard for Cable Telephony

This paper presents the BroadVoice16 (BV16) speech codec, which is a mandatory codec in the PacketCable 1.5 standard. For cable telephony based on PacketCable 1.5, BV16 possesses a set of attributes not met by other speech codecs: (1) no royalty, (2) high quality, (3) low delay, (4) low complexity, and (5) medium to low bit-rate. The royalty-free requirement excludes many modern speech coding techniques. Hence, an older paradigm is resurrected and improved as the foundation of BV16. Extensive test results including independent subjective tests, PESQ evaluation across 13 languages, and DTMF pass-through evaluation demonstrate the high performance of BV16.

Analysis-by-Synthesis Speech Coding

Since the early 1980s, advances in speech coding technologies have enabled speech coders to achieve bit-rate reductions of a factor of 4 to 8 while maintaining roughly the same high speech quality. One of the most important driving forces behind this feat is the so-called analysis-by-synthesis paradigm for coding the excitation signal of predictive speech coders. In this chapter, we give an overview of many variations of the analysis-by-synthesis excitation coding paradigm as exemplified by various speech coding standards around the world. We describe the variations of the same basic theme in the context of different coder structures where these techniques are employed. We also attempt to show the relationship between them in the form of a family tree. The goal of this chapter is to give the readers a big-picture understanding of the dominant types of analysis-by-synthesis excitation coding techniques for predictive speech coding.

System architectures and digital signal processing algorithms for enhancing the output audio quality of stereo FM broadcast receivers

This paper presents two FM receiver architectures and three digital signal processing algorithms for enhancing the output audio quality of FM broadcast receivers. The two receiver architectures differ only in the front-end processing for estimating the carrier-to-noise ratio and noise floors of the stereo audio signals. The shared back-end processing consists of three algorithms to suppress the noise in the audio signal, to detect and cancel noise pulses (static), and to conceal the degrading effects of fast fading, respectively. Together these FM enhancement techniques achieve about 20 to 35 dB improvements in SNR and stereo separation over a wide range of RF signal strength spanning nearly 30 dB. Perceptually, the audio quality improvement is large and obvious when the received FM signal is weak.