Erdal Paksoy

A variable rate multimodal speech coder with gain-matched analysis-by-synthesis

In general, a variable rate coder can obtain the same speech quality as a fixed rate coder, while reducing the average bit rate. We have developed a variable-rate multimodal speech coder with an average bit rate of 3 kb/s for a speech activity factor of 80% and quality comparable to the GSM full rate coder. The coder has four coding modes and uses a robust classification method involving the pitch gain, zero crossings, and a peakiness measure. Also the coder employs a novel gain-matched analysis-by-synthesis technique for very low rate coding of unvoiced frames and an improved noise-level-dependent postfilter. This paper describes the details of our algorithm and presents the results from subjective listening tests.

An adaptive multi-rate speech coder for digital cellular telephony

We have developed an adaptive multi-rate (AMR) speech coder designed to operate under the GSM digital cellular full rate (22.8 kb/s) and half rate (11.4 kb/s) channels and to maintain high quality in the presence of highly varying background noise and channel conditions. Within each total rate, several codec modes with different source/channel bit rate allocations are used. The speech coders in each codec mode are based on the CELP algorithm operating at rates ranging from 11.85 kb/s down to 5.15 kb/s, where the lowest rate coder is a source controlled multi-modal speech coder. The decoders monitor the channel quality at both ends of the wireless link using the soft values for the received bits and assist the base station in selecting the codec mode that is appropriate for a given channel condition. The coder was submitted to the GSM AMR standardization competition and met the qualification requirements in an independent formal MOS test.

A 4 kb/s hybrid MELP/CELP speech coding candidate for ITU standardization

This paper presents an improved 4 kb/s hybrid MELP/CELP speech coder submitted as a candidate for ITU standardization. The coder uses three modes: a high-quality MELP coder for strongly voiced speech frames, an ACELP coder with pitch prediction for weakly voiced frames, and a stochastic CELP coder for unvoiced frames. We present recent enhancements to this coder, both to improve speech quality and to reduce coder complexity. Previous lTU Selection Testing results on an earlier version of this coder showed that it met nearly all requirements for toll-quality speech, more than any other candidate. Our internal testing shows that the current reduced-complexity fixed-point coder maintains this high performance.

An adaptive voice playout method for VOP applications

We propose an adaptive voice playout method for handling network delay jitter in voice over packet (VOP) receivers. Our method allows playout delay increase during both silence periods and active speech; however, it allows playout delay decrease during silence periods only. Since the playout delay is increased during active speech, a speech extension algorithm is required. Therefore we have developed a novel, voicing-classification based speech extension algorithm for CELP speech coders. Though the complementary speech truncation algorithm is not needed for our adaptive playout mechanism, it is suitable for applications such as voice synchronization with other media.

A variable-rate CELP coder for fast remote voicemail retrieval using a notebook computer

Remote retrieval of compressed voicemail data over a telephone line is one of several emerging applications of speech coding. Using a notebook computer equipped with a modem, a user can remotely access a networked desktop unit located at their office or home to retrieve various types of information such as email, FAX, electronic documents as well as voicemail. By compressing the speech data, we reduce the amount of time needed for the transfer of voice data over the telephone line. We have developed a high-quality variable-rate CELP (code-excited linear prediction) coder which can be used in such an application. The coder operates at an average rate of 3 kb/s assuming 80% speech activity and uses several new techniques including a mode decision mechanism based on the use of a peakiness measure and a novel excitation search method called gain-matched analysis-by-synthesis. The coder gives quality comparable to the Microsoft GSM 6.10 Audio Codec at 13 kb/s.