The past three decades has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for voice applications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. Then I proceed with descriptions of waveform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. At the end the system for computer-based stenographing is presented. Quality research and ways how to improve this system will be provided.

<p>The voice channel in mobile communication systems have high priority and are almost always available. By using the voice channel also for data transmissions it is possible to get the same availability as for voice calls. But due to speech codecs in the voice channel, regular modems can not be used and special techniques are needed to transmit data.</p><p>This thesis presents methods to transmit data over the voice channel in a GSM, UMTS or TETRA network. The focus has been on robust data transmission rather than high data bit rates. Approaches are introduced which improve the reliability for transmissions even for systems with low rate speech codecs and channels with some distortion.</p><p>The results of the thesis are suggestions of symbol patterns and ways to create and adapt symbols for specific application and channel conditions to achieve the desired goal for the application.</p>

The voice channel in mobile communication systems have high priority and are almost always available. By using the voice channel also for data transmissions it is possible to get the same availability as for voice calls. But due to speech codecs in the voice channel, regular modems can not be used and special techniques are needed to transmit data. This thesis presents methods to transmit data over the voice channel in a GSM, UMTS or TETRA network. The focus has been on robust data transmission rather than high data bit rates. Approaches are introduced which improve the reliability for transmissions even for systems with low rate speech codecs and channels with some distortion. The results of the thesis are suggestions of symbol patterns and ways to create and adapt symbols for specific application and channel conditions to achieve the desired goal for the application.

<p>Todays’ systems for telecommunication are getting more and more complex. Automatic testing is required to guarantee quality of the systems produced. An actual example is the introduction of GPRS traffic in the GSM network nodes. This thesis investigates the need and demands for such an automatic testing of the traffic channels in the GSM system. A solution intended to be a part of the Ericsson TSS is proposed. One problem to be solved is that today’s tools for testing do not support testing of speech channels with the speech transcoder unit installed. As part of the investigation, a speech codec is implemented for execution on current hardware used in the test platform. The selected speech codec is the enhanced full rate codec, generating a bitstream of 12.2 kbit/s, and gives a good trade-off between compression and speech quality. The report covers the design of the test tool and the implementation of speech codec. Particularly performance problems in the implementation of the encoder will be addressed.</p>

Todays’ systems for telecommunication are getting more and more complex. Automatic testing is required to guarantee quality of the systems produced. An actual example is the introduction of GPRS traffic in the GSM network nodes. This thesis investigates the need and demands for such an automatic testing of the traffic channels in the GSM system. A solution intended to be a part of the Ericsson TSS is proposed. One problem to be solved is that today’s tools for testing do not support testing of speech channels with the speech transcoder unit installed. As part of the investigation, a speech codec is implemented for execution on current hardware used in the test platform. The selected speech codec is the enhanced full rate codec, generating a bitstream of 12.2 kbit/s, and gives a good trade-off between compression and speech quality. The report covers the design of the test tool and the implementation of speech codec. Particularly performance problems in the implementation of the encoder will be addressed.