Low Bit Rate Speech Coding Using TMS320C6416
The title of the project is Low Bit Rate Speech Coding Using TMS320C6416 DSP Processor. The scope of this project is divided into two main parts. Part one involves the study of the TMS320C6416 DSP processor. My task was to understand the architecture of this board and complete the tutorials in...
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| Format: | Monograph |
| Language: | English |
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Universiti Sains Malaysia
2005
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| Online Access: | http://eprints.usm.my/57737/ http://eprints.usm.my/57737/1/Low%20Bit%20Rate%20Speech%20Coding%20Using%20TMS320C6416_Mohamad%20Habib%20Mahamad%20Haniffah.pdf |
| _version_ | 1848883705120030720 |
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| author | Mahamad Haniffah, Mohamad Habib |
| author_facet | Mahamad Haniffah, Mohamad Habib |
| author_sort | Mahamad Haniffah, Mohamad Habib |
| building | USM Institutional Repository |
| collection | Online Access |
| description | The title of the project is Low Bit Rate Speech Coding Using TMS320C6416 DSP
Processor. The scope of this project is divided into two main parts. Part one involves the
study of the TMS320C6416 DSP processor. My task was to understand the architecture
of this board and complete the tutorials in Code Composer Studio (CCS). The second part
is concerned with the sampling of speech signal (analog signal) at different sampling
frequencies and to study its effects on the quality of the reconstructed speech signal.
Initially MATLAB and SIMULINK were used to sample the speech file and to study the
effect of variation in sampling frequency on the quality of the speech signal and its
waveform. Later, the sampling process is implemented in real time using the
TMS320C6416 DSP Processor. Three sampling frequencies were chosen which are 8000
Hz, 4000 Hz and 2000 Hz. The results were divided into two sections; before real-time
implementation and after real-time implementation. The comparison of the quality of the
sampled audio signal was carried out for the three sampling frequencies as mentioned
earlier. Two methods were used to measure the quality of the reconstructed audio signal.
First, fifteen students were chosen to rate their score for the quality of the reconstructed
signal. The score range was from 1(bad) to 5(excellent). Secondly, scope was used to
display the waveform of the original and reconstructed signal. The results showed that the
quality of the sound degrades from 8000 Hz to 2000 Hz. |
| first_indexed | 2025-11-15T18:55:03Z |
| format | Monograph |
| id | usm-57737 |
| institution | Universiti Sains Malaysia |
| institution_category | Local University |
| language | English |
| last_indexed | 2025-11-15T18:55:03Z |
| publishDate | 2005 |
| publisher | Universiti Sains Malaysia |
| recordtype | eprints |
| repository_type | Digital Repository |
| spelling | usm-577372023-04-03T06:28:40Z http://eprints.usm.my/57737/ Low Bit Rate Speech Coding Using TMS320C6416 Mahamad Haniffah, Mohamad Habib T Technology TK Electrical Engineering. Electronics. Nuclear Engineering The title of the project is Low Bit Rate Speech Coding Using TMS320C6416 DSP Processor. The scope of this project is divided into two main parts. Part one involves the study of the TMS320C6416 DSP processor. My task was to understand the architecture of this board and complete the tutorials in Code Composer Studio (CCS). The second part is concerned with the sampling of speech signal (analog signal) at different sampling frequencies and to study its effects on the quality of the reconstructed speech signal. Initially MATLAB and SIMULINK were used to sample the speech file and to study the effect of variation in sampling frequency on the quality of the speech signal and its waveform. Later, the sampling process is implemented in real time using the TMS320C6416 DSP Processor. Three sampling frequencies were chosen which are 8000 Hz, 4000 Hz and 2000 Hz. The results were divided into two sections; before real-time implementation and after real-time implementation. The comparison of the quality of the sampled audio signal was carried out for the three sampling frequencies as mentioned earlier. Two methods were used to measure the quality of the reconstructed audio signal. First, fifteen students were chosen to rate their score for the quality of the reconstructed signal. The score range was from 1(bad) to 5(excellent). Secondly, scope was used to display the waveform of the original and reconstructed signal. The results showed that the quality of the sound degrades from 8000 Hz to 2000 Hz. Universiti Sains Malaysia 2005-03-01 Monograph NonPeerReviewed application/pdf en http://eprints.usm.my/57737/1/Low%20Bit%20Rate%20Speech%20Coding%20Using%20TMS320C6416_Mohamad%20Habib%20Mahamad%20Haniffah.pdf Mahamad Haniffah, Mohamad Habib (2005) Low Bit Rate Speech Coding Using TMS320C6416. Project Report. Universiti Sains Malaysia, Pusat Pengajian Kejuruteraan Elektrik dan Elektronik. (Submitted) |
| spellingShingle | T Technology TK Electrical Engineering. Electronics. Nuclear Engineering Mahamad Haniffah, Mohamad Habib Low Bit Rate Speech Coding Using TMS320C6416 |
| title | Low Bit Rate Speech Coding Using TMS320C6416 |
| title_full | Low Bit Rate Speech Coding Using TMS320C6416 |
| title_fullStr | Low Bit Rate Speech Coding Using TMS320C6416 |
| title_full_unstemmed | Low Bit Rate Speech Coding Using TMS320C6416 |
| title_short | Low Bit Rate Speech Coding Using TMS320C6416 |
| title_sort | low bit rate speech coding using tms320c6416 |
| topic | T Technology TK Electrical Engineering. Electronics. Nuclear Engineering |
| url | http://eprints.usm.my/57737/ http://eprints.usm.my/57737/1/Low%20Bit%20Rate%20Speech%20Coding%20Using%20TMS320C6416_Mohamad%20Habib%20Mahamad%20Haniffah.pdf |