In this paper, we develop a 3D audio reproduction scheme for the purpose of delivering audio over IP networks. In this
scheme, audio streams constructed at a server are composed of the TCP/IP header followed by multi-channel audio data
compressed by MPEG advanced audio coding (AAC), and the decoded audio signals are played out on a stereo
loudspeaker system at the client. Since the audio source is recorded by a multi-channel microphone but the playout is
dedicated to stereo speakers, the quality mismatch between the multi-channel and the stereo system should be overcome.
As a potential solution, we first investigate the effect of 3D audio processing on the audio quality at the client by
applying a head-related transfer function (HRTF). Next, a crosstalk cancellation process is applied to the audio with 3D
effects in order to improve the immersion of the processed 3D effects on a stereo loudspeaker system. Finally, we
evaluate the performance of the 3D audio reproduction system in terms of the identification of an audio source and
quality comparison before and after applying the crosstalk cancellation technique.
Proc. SPIE. 6777, Multimedia Systems and Applications X
KEYWORDS: Digital signal processing, Telecommunications, Signal processing, Computer programming, Control systems, Data communications, Quantization, Electronic filtering, Televisions, Error control coding
In this paper, we design and implement a two-way real-time communication system for audio over cable television
(CATV) networks to provide an audio-based interaction between the CATV broadcasting station and CATV subscribers.
The two-way real-time communication system consists of a real-time audio encoding/decoding module, a payload
formatter based on a transmission control protocol/Internet protocol (TCP/IP), and a cable network. At the broadcasting
station, audio signals from a microphone are encoded by an audio codec that is implemented using a digital signal
processor (DSP), where the MPEG-2 Layer II audio codec is used for the audio codec and TMS320C6416 is used for a
DSP. Next, a payload formatter constructs a TCP/IP packet from an audio bitstream for transmission to a cable modem.
Another payload formatter at the subscriber unpacks the TCP/IP packet decoded from the cable modem into audio
bitstream. This bitstream is decoded by the MPEG-2 Layer II audio decoder. Finally the decoded audio signals are
played out to the speaker. We confirmed that the system worked in real-time, with a measured delay of around 150 ms
including the algorithmic and processing time delays.
In this paper, we investigate the use of existing audio codecs for the purpose of a high quality color ring-back-
tone service. First of all, we exploit the limitations of the enhanced variable rate codec (EVRC) in a view of
music quality because EVRC is a standard speech coder employed in a code division multiple access (CDMA)
system. In order to figure it out which current existing audio codec is suitable to deliver music over CDMA
or wideband CDMA (W-CDMA), several audio codecs such as two different versions of MPEG AAC and the
Enhanced AAC+ codec are reviewed. Next, the music quality of the audio codecs is compared with that of
EVRC, where the bit-rates of the audio codecs are set to be around 10 kbit/s because the color ring-back-tone
service using one of the audio codecs should be realized by replacing EVRC with it. The quality comparison is
performed by an informal listening test as well as an objective quality test. It is shown from the experiments
that the audio codecs provide better music quality than EVRC and among them, the Enhance AAC+ codec
operated at a bit-rate of 10 kbit/s with a sampling rate of 32 kHz can be considered as a new candidate for the
high quality color ring-back-tone service.