ﺑﺎﺳﻤﻪ ﺗﻌﺎﻟﯽ ﺳﯿﺴﺘﻢ ﻫﺎي ﭼﻨﺪرﺳﺎﻧﻪاي )(40-342 داﻧﺸﮑﺪه ﻣﻬﻨﺪﺳﯽ ﮐﺎﻣﭙﯿﻮﺗﺮ ﺗﺮم ﺑﻬﺎر 1387 دﮐﺘﺮ ﺣﻤﯿﺪرﺿﺎ رﺑﯿﻌﯽ ﺗﮑﻠﯿﻒ ﺷﻤﺎره :2ﻓﺸﺮده ﺳﺎزي ﺳﯿﮕﻨﺎل :ﮔﻔﺘﺎر و ﺻﻮت -1ﻣﻘﺪﻣﻪ ﺁﻧﭽﻪ ﺗﻮﺯﻳﻊ ﺳﻴﮕﻨﺎﻟﻬﺎﻱ ﺻﻮﺕ ﻭ ﮔﻔﺘﺎﺭ ﺭﺍ ﺑﺪﻭﻥ ﻧﻴﺎﺯ ﺑﻪ ﺍﺧﺘﺼﺎﺹ ﭘﻬﻨﺎﻱ ﺑﺎﻧﺪ ﻭﺳﻴﻌﻲ ﺑﺮﺍﻱ ﺍﻧﺘﻘﺎﻝ ﻭ ﺫﺧﻴﺮﻩ ﺣﺠﻢ ﺯﻳﺎﺩﻱ ﺍﺯ ﺳﻴﮕﻨﺎﻟﻬﺎﻱ ﺻﻮﺗﻲ ﻣﻤﻜﻦ ﻣﻲ ﺳﺎﺯﺩ ،ﺗﻜﻨﻴﻚ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﻳﺎ ﺑﻪ ﺑﻴﺎﻥ ﺩﻳﮕﺮ ﻛﺪ ﻛﺮﺩﻥ ﺍﺳﺖ .ﺍﻳﻦ ﺗﻜﻨﻴﻚ ﻣﻘﺪﺍﺭ ﺩﺍﺩﻩ ﻣﻮﺭﺩﻧﻴﺎﺯ ﺑﺮﺍﻱ ﺍﻧﺘﻘﺎﻝ ﻭ ﺫﺧﻴﺮﻩ ﺻﻮﺕ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺷﺪﻩ ﺩﻳﺠﻴﺘﺎﻟﻲ ﺭﺍ ﻫﻢ ﺩﺭ ﻃﻮﻝ ﻣﺮﺣﻠﻪ ﺗﺒﺪﻳﻞ ﺁﻧﺎﻟﻮﮒ – ﺑﻪ – ﺩﻳﺠﻴﺘﺎﻝ ﻭ ﻫﻢ ﭘﺲ ﺍﺯ ﺫﺧﻴﺮﻩ ﻓﺎﻳﻞ ﺧﺎﻡ ﺑﻪ ﺻﻮﺭﺕ ﺩﻳﺠﻴﺘﺎﻟﻲ ،ﻛﺎﻫﺶ ﻣﻲ ﺩﻫﺪ .ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﻭ ﻋﻜﺲ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﺑﻮﺳﻴﻠﻪ ﺍﻟﮕﻮﺭﻳﺘﻤﻬﺎﻱ ﻣﺘﻌﺪﺩﻱ ﻗﺎﺑﻞ ﭘﻴﺎﺩﻩ ﺳﺎﺯﻱ ﺍﺳﺖ ﻛﻪ ﻣﻲ ﺗﻮﺍﻧﺪ ﺩﺭ ﻛﺎﺭﺑﺮﺩﻫﺎﻱ ﻧﺮﻡ ﺍﻓﺰﺍﺭﻱ ﻳﺎ ﻛﺎﺭﺑﺮﺩﻫﺎﻱ ﺧﺎﺹ ﻣﺪﺍﺭﻫﺎﻱ ﻣﺠﺘﻤﻊ )ﺗﺮﺍﺷﻪ ﻫﺎ( ﺑﻜﺎﺭ ﮔﺮﻓﺘﻪ ﺷﻮﺩ. ﺍﺳﺘﺎﻧﺪﺍﺭﺩﻫﺎﻱ ﻣﺘﻌﺪﺩ ﺟﻬﺎﻧﻲ ﺑﺮﺍﻱ ﻛﺪ ﻛﺮﺩﻥ ﻭﻳﺪﻳﻮ ﻭ ﺻﻮﺕ ﭘﺎﻳﻪ ﺭﻳﺰﻱ ﺷﺪﻩ ﺍﻧﺪ .ﺑﺮﺧﯽ ﺍﺯ ﺍﻳﻦ ﺍﺳﺘﺎﻧﺪﺍﺭﺩﻫﺎ ﺷﺎﻣﻞ MPEG-،MPEG-1 2ﻭ MPEG-4ﻣﻲ ﺑﺎﺷﻨﺪ .ﺍﻟﺒﺘﻪ ﺍﺳﺘﺎﻧﺪﺍﺭﺩﻫﺎﻱ ﻣﺘﻌﺪﺩﻱ ﺑﺮﺍﻱ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﻭ ﻋﻜﺲ ﻋﻤﻞ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﺷﻜﻞ ﻣﻮﺟﻬﺎﻱ ﺻﻮﺕ ﻭ ﮔﻔﺘﺎﺭ ﺑﺮﺍﻱ ﻛﺎﺭﺑﺮﺩﻫﺎﻱ desktopﭼﻨﺪ ﺭﺳﺎﻧﻪ ﺍﻱ ﻭﺟﻮﺩ ﺩﺍﺭﺩ .ﺑﺨﺶ ﻫﺎﻱ ﺑﻌﺪﻱ ﺑﻪ ﺍﻟﮕﻮﺭﻳﺘﻤﻬﺎﻱ ﻣﻌﻤﻮﻝ ﻭ ﺍﻧﻮﺍﻉ ﮔﻮﻧﺎﮔﻮﻧﻲ ﺍﺯ ﺭﻭﺷﻬﺎﻱ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﺑﺮﺍﻱ ﺻﻮﺕ ﻭ ﮔﻔﺘﺎﺭ ﺍﺧﺘﺼﺎﺹ ﺩﺍﺭﺩ. -2ﺗﺌﻮرﯾﻬﺎ و ﻃﺮﺣﻬﺎ ﻣﺎ ﻗﺒﻼﹰ ﺗﺌﻮﺭﻱ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺭﺍ ﺩﺭ ﺗﻜﻠﻴﻒ ١ﻣﻮﺭﺩ ﺑﺮﺭﺳﻲ ﻗﺮﺍﺭ ﺩﺍﺩﻩ ﺍﻳﻢ .ﻫﻤﭽﻨﻴﻦ ﻧﺸﺎﻥ ﺩﺍﺩﻳﻢ ﻛﻪ ﻧﻤﻮﻧﻪ ﻫﺎﻱ ﻳﻚ ﺳﻴﮕﻨﺎﻝ ﺁﻧﺎﻟﻮﮒ ﻧﻤﺎﻳﺶ ﻣﻨﺤﺼﺮ ﺑﻪ ﻓﺮﺩﻱ ﺍﺯ ﺳﻴﮕﻨﺎﻝ ﺍﻭﻟﻴﻪ ﻣﻲ ﺑﺎﺷﻨﺪ ﺑﻪ ﺷﺮﻃﻲ ﻛﻪ ﭘﻬﻨﺎﻱ ﺑﺎﻧﺪ ﻣﺤﺪﻭﺩ ﺩﺍﺷﺘﻪ ﺑﺎﺷﻨﺪ ﻭ ﻧﺮﺥ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺣﺪﺍﻗﻞ ﺩﻭﺑﺮﺍﺑﺮ ﻓﺮﻛﺎﻧﺲ ﺳﻴﮕﻨﺎﻝ ﺑﺎﺷﺪ .ﺍﺯ ﺁﻧﺠﺎﻳﻲ ﻛﻪ ﻣﺎ ﺑﺎ ﻧﻤﺎﻳﺶ ﺩﻳﺠﻴﺘﺎﻟﻲ ﺻﻮﺕ ﻭ ﮔﻔﺘﺎﺭ ﺳﺮ ﻭ ﻛﺎﺭ ﺩﺍﺭﻳﻢ ،ﺍﺣﺘﻴﺎﺝ ﺑﻪ ﺩﺍﻧﺴﺘﻦ ﺧﺼﻮﺻﻴﺎﺕ ﻃﻴﻔﻲ ﺻﺪﺍ ﻭ ﮔﻔﺘﺎﺭ ﺩﺍﺭﻳﻢ .ﺑﻪ ﺭﺍﺣﺘﻲ ﻣﺸﺎﻫﺪﻩ ﻣﻲ ﺷﻮﺩ ﻛﻪ ﺑﺮﺍﻱ ﺻﺪﺍﻫﺎﻱ ﻭﺍﮐﺪﺍﺭ ،ﺩﺍﻣﻨﻪ ﻃﻴﻒ ﻓﺮﻛﺎﻧﺴﻲ ﺳﻴﮕﻨﺎﻝ ﺩﺭ ﻓﺮﻛﺎﻧﺴﻬﺎﻱ ﺑﺎﻻﻱ ٤٠dB ،٤KHzﭘﺎﻳﻴﻦ ﺗﺮ ﺍﺯ ﻗﻠﻪ ﻃﻴﻔﻲ ﺳﻴﮕﻨﺎﻝ ﺍﺳﺖ .ﺍﺯ ﻃﺮﻑ ﺩﻳﮕﺮ ،ﺩﺭ ﺳﻴﮕﻨﺎﻟﻬﺎﻱ ﺻﻮﺗﻲ ،ﻃﻴﻒ ﺳﻴﮕﻨﺎﻝ ﺣﺘﻲ ﺩﺭ ﻓﺮﻛﺎﻧﺴﻬﺎﻱ ﺑﺎﻻﻱ ٨KHzﺑﻪ ﻃﻮﺭ ﻗﺎﺑﻞ ﻣﻼﺣﻈﻪ ﺍﻱ ﺍﻓﺖ ﻧﻤﻲ ﻛﻨﺪ .ﻋﻼﻭﻩ ﺑﺮ ﺍﻳﻦ ،ﺩﺭ ﻛﺎﺭﺑﺮﺩﻫﺎﻱ ﻛﺎﻣﭙﻴﻮﺗﺮﻱ ﺑﺮﺍﻱ ﻧﻤﺎﻳﺶ ﺳﻴﮕﻨﺎﻝ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺷﺪﻩ ،ﻣﻘﺎﺩﻳﺮ ﻣﻤﻜﻦ ﻳﻚ ﻧﻤﻮﻧﻪ ﻛﻪ ﺩﺭ ﻣﺤﺪﻭﺩﺓ ﭘﻴﻮﺳﺘﻪ ﺍﻱ ﺗﻐﻴﻴﺮ ﻣﻲ ﻛﻨﻨﺪ ﺑﺎﻳﺪ ﺑﻪ ﺗﻌﺪﺍﺩ ﻣﺤﺪﻭﺩﻱ ﺍﺯ ﻣﻘﺎﺩﻳﺮ ﮔﺴﺴﺘﻪ ﺗﺒﺪﻳﻞ ﺷﻮﺩ .ﺍﻳﻦ ﭘﺮﻭﺳﻪ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻧﺎﻣﻴﺪﻩ ﻣﻲ ﺷﻮﺩ. -1-2ﮐﻮاﻧﺘﯿﺰاﺳﯿﻮن ﺳﯿﮕﻨﺎﻟﻬﺎي ﻧﻤﻮﻧﻪ ﺑﺮداري ﺷﺪه ﻣﺤﺪﻭﺩﻩ ﻫﺎ ﻭ ﺳﻄﻮﺡ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻣﻤﻜﻦ ﺍﺳﺖ ﺑﻪ ﺻﻮﺭﺗﻬﺎﻱ ﻣﺘﻌﺪﺩﻱ ﺍﻧﺘﺨﺎﺏ ﺷﻮﻧﺪ ﻛﻪ ﺑﺴﺘﮕﻲ ﺑﻪ ﻛﺎﺭﺑﺮﺩﻫﺎﻱ ﺍﺯ ﭘﻴﺶ ﺗﻌﻴﻴﻦ ﺷﺪﺓ ﻧﻤﺎﻳﺶ ﺩﻳﺠﻴﺘﺎﻟﻲ ﺁﻥ ﺩﺍﺭﺩ .ﺑﺎ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ،ﻣﺤﺪﻭﺩﺓ ﺩﻳﻨﺎﻣﻴﻚ )ﺣﺪﺍﻗﻞ ﺗﺎ ﺣﺪﺍﻛﺜﺮ( ﺳﻴﮕﻨﺎﻝ ،Rﺑﻪ Wﺑﺎﺯﻩ ﺑﺎ ﻃﻮﻝ ﻳﻜﺴﺎﻥ D ﺗﻘﺴﻴﻢ ﻣﻲ ﺷﻮﺩ .ﻣﺎ Dﺭﺍ ﭘﻠﺔ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻣﻲ ﻧﺎﻣﻴﻢ .ﺭﺍﺑﻄﻪ ﻭﺭﻭﺩﻱ ﻭ ﻣﻘﺪﺍﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻧﺸﺪﻩ ،ﻭ ﺧﺮﻭﺟﻲ )ﻣﻘﺪﺍﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ( ﺑﺮﺍﻱ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ﺩﺭ ﺷﮑﻞ ١ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﺷﺪﻩ ﺍﺳﺖ .ﻛﻪ ﺩﺭ ﺁﻥ ،xiﻣﺤﺪﻭﺩﺓ ﺭﺍﺳﺖ ﺑﺎﺯﺓ iﻭ xiﺳﻄﺢ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﺓ ﺍﻳﻦ ﺑﺎﺯﻩ ﺍﺳﺖ ﻛﻪ ﺷﺮﻁ ﻫﺎﻱ ﺯﻳﺮ ﺭﺍ ﺑﺮﺁﻭﺭﺩﻩ ﻣﻲ ﺳﺎﺯﺩ. )(١-٣ 1 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 )(٢-٣ ﻫﺮ ﻣﻘﺪﺍﺭ ﺩﺭ ﻣﺤﺪﻭﺩﺓ iﺍﻡ ﺑﻪ ﻣﻘﺪﺍﺭ ﻣﻴﺎﻧﻲ ﺍﻳﻦ ﻣﺤﺪﻭﺩﻩ ﻧﮕﺎﺷﺖ ﻣﻲﺷﻮﺩ. )(٣-٣ ﺩﺭ ﻛﺎﻣﭙﻴﻮﺗﺮ ،ﻫﺮ ﺳﻄﺢ ﺑﺎ ﻳﻚ ﻛﻠﻤﻪ ﻛﺪ ﺑﺎﻳﻨﺮﻱ ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﻣﻲ ﺷﻮﺩ .ﺑﺎ Wﺳﻄﺢ ﻛﻮﺍﻧﻴﺰ ﺷﺪﻩ ،ﻫﺮ ﺳﻄﺢ ﻣﻲﺗﻮﺍﻧﺪ ﺑﺎ ]) B = [log2(Lﺑﻴﺖ ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﺷﻮﺩ )ﺷﮑﻞ.(١ ﺷﮑﻞ -1ﺧﺼﻮﺻﯿﺎت ورودي – ﺧﺮوﺟﯽ ﯾﮏ ﮐﻮاﻧﯿﺰه ﮐﻨﻨﺪة 3ﺑﯿﺘﯽ ﺍﮔﺮ ﻣﺤﺪﻭﺩﺓ ﺳﻴﮕﻨﺎﻝ Rﺑﺎﺷﺪ ،ﻳﻚ ﻛﻮﺍﻧﻴﺰﻩ ﻛﻨﻨﺪﻩ ﻳﻜﻨﻮﺍﺧﺖ ﻓﻘﻂ ﻳﻚ ﭘﺎﺭﺍﻣﺘﺮ ﺩﺍﺭﺩ :ﺗﻌﺪﺍﺩ ﺳﻄﻮﺡ Nﻳﺎ ﺍﻧﺪﺍﺯﻩ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ، Dﻛﻪ ﻫﺮ ﺩﻭ ﺑﺎ ﺭﺍﺑﻄﺔ ﺯﻳﺮ ﺑﻪ ﻫﻢ ﺍﺭﺗﺒﺎﻁ ﺩﺍﺭﻧﺪ. )(٤-٣ B ﺗﻌﺪﺍﺩ ﺳﻄﻮﺡ Nﻣﻌﻤﻮﻻﹰ ﺑﻪ ﮔﻮﻧﻪ ﺍﻱ ﺍﻧﺘﺨﺎﺏ ﻣﻲ ﺷﻮﻧﺪ ﻛﻪ ﺑﻪ ﺻﻮﺭﺕ 2ﺑﺎﺷﻨﺪ ﺗﺎ ﺑﻬﺘﺮﻳﻦ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﻛﻠﻤﺔ ﻛﺪ Bﺑﻴﺘﻲ ﺷﻮﺩ .ﺍﮔﺮ ﺳﻴﮕﻨﺎﻝ ﺗﺎﺑﻊ ﭼﮕﺎﻟﻲ ﺍﺣﺘﻤﺎﻝ ﻣﺘﻘﺎﺭﻥ ﺑﺎﺷﺪ ﺑﻪ ﺍﻳﻦ ﺗﺮﺗﻴﺐ ﻛﻪ | x (n ) |£ x maxﻳﺎ R=2xmaxﺑﺎﺷﺪ،ﺁﻧﮕﺎﻩ ﺑﺎﻳﺪ ﻣﻘﺎﺩﻳﺮ ﺯﻳﺮ ﺗﻨﻈﻴﻢ ﺷﻮﻧﺪ. )(٥-٣ ﺩﺭ ﺑﺤﺚ ﺗﺎﺛﻴﺮﺍﺕ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻣﻔﻴﺪ ﺑﻪ ﻧﻈﺮ ﻣﻲ ﺭﺳﺪ ﻛﻪ ﻣﻘﺎﺩﻳﺮ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺷﺪﻩ ) x(nﺭﺍ ﺑﻪ ﺻﻮﺭﺕ ﺯﻳﺮ ﻧﻤﺎﻳﺶ ﺩﻫﻴﻢ )(٦-٣ ﻛﻪ ) x(nﻧﻤﻮﻧﻪ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻧﺸﺪﻩ e(n) ،ﺧﻄﺎﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻳﺎ ﻧﻮﻳﺰ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺍﺳﺖ .ﺍﺯ ﺷﻜﻞ ١ﺩﻳﺪﻩ ﻣﻲ ﺷﻮﺩ ﻛﻪ ﺍﮔﺮ Dﻭ Bﻣﺎﻧﻨﺪ ﺭﺍﺑﻄﺔ )(٥-٣ ﺍﻧﺘﺨﺎﺏ ﺷﻮﻧﺪ ،ﺁﻧﮕﺎﻩ )(٧-٣ ﻧﺴﺒﺖ ﺳﻴﮕﻨﺎﻝ ﺑﻪ ﻧﻮﻳﺰ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺩﺭ ﺭﺍﺑﻄﻪ ﺯﻳﺮ ﺑﻴﺎﻥ ﺷﺪﻩ ﺍﺳﺖ. 2 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 )(٨-٣ R2 ﺑﻪ ﻳﺎﺩ ﺑﻴﺎﻭﺭﻳﻢ ﻛﻪ ﺑﺮﺍﻱ ﺳﻴﮕﻨﺎﻝ ﺑﺎ ﺗﻮﺯﻳﻊ ﻳﻜﻨﻮﺍﺧﺖ ﺩﺭ ﻣﺤﺪﻭﺩﺓ ،Rﻭﺍﺭﻳﺎﻧﺲ ﺑﺮﺍﺑﺮ 12 D D ) (- ,ﻳﻜﻨﻮﺍﺧﺖ ﺑﺎﺷﺪ ،ﺭﺍﺑﻄﻪ ﺯﻳﺮ ﺑﺮﺍﻱ ﻧﻮﻳﺰ ﻧﺘﻴﺠﻪ ﻣﻲ ﺷﻮﺩ. 2 2 ﺍﺳﺖ .ﺍﮔﺮ ﺗﻮﺯﻳﻊ ﺩﺍﻣﻨﻪ ﻧﻮﻳﺰ ﺩﺭ ﺑﺎﺯﺓ )(۹-٣ ﺑﺎ ﺟﺎﻳﮕﺰﻳﻨﻲ ﺭﺍﺑﻄﺔ ) (٩-٣ﺩﺭ ﺭﺍﺑﻄﺔ ):(٨-٣ )(١٠-٣ ﻳﺎ ﺑﻴﺎﻥ ﺧﻄﺎﻱ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﺩﺭ ﻭﺍﺣﺪ dB )(۱۱-۳ ﺍﮔﺮ ﻣﺤﺪﻭﺩﺓ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﺭﺍ xmax = 4s xﻓﺮﺽ ﻛﻨﻴﻢ ﺳﭙﺲ ﺭﺍﺑﻄﺔ ) (١١-٣ﺑﻪ ﺻﻮﺭﺕ ﺯﻳﺮ ﺩﺭ ﻣﻲ ﺁﻳﺪ. )(١٢-٣ ﺍﻳﻦ ﺭﺍﺑﻄﻪ ﺑﻴﺎﻥ ﻣﻲ ﻛﻨﺪ ﻛﻪ ﻫﺮ ﺑﻴﺖ ﺍﺿﺎﻓﻲ 6dB ،ﺑﻪ ﺑﻬﺒﻮﺩ SNRﻛﻤﻚ ﻣﻲ ﻛﻨﺪ .ﺑﺮﺍﻱ ﺣﻔﻆ ﺍﻋﺘﺒﺎﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﺮﺩﻥ ﻳﻜﻨﻮﺍﺧﺖ ،ﻻﺯﻡ ﺍﺳﺖ ﺗﺎ ﺗﻌﺪﺍﺩ ﺑﻴﺖ ﺑﻴﺸﺘﺮﻱ ﻧﺴﺒﺖ ﺑﻪ ﺁﻧﺎﻟﻴﺰ ﻗﺒﻠﻲ ﻛﻪ ﺩﺭ ﺁﻥ ﺳﻴﮕﻨﺎﻝ ﺍﻳﺴﺘﺎﻥ ﻭ ﺩﺍﺭﺍﻱ ﺗﻮﺯﻳﻊ ﻣﺘﻘﺎﺭﻥ ﻓﺮﺽ ﻣﻲ ﺷﺪ ﻭ X max = 4s xﺑﻮﺩ، ﺍﺧﺘﺼﺎﺹ ﻳﺎﺑﺪ .ﺑﻪ ﻋﻨﻮﺍﻥ ﻣﺜﺎﻝ ،ﺩﺭ ﺣﺎﻟﻲ ﻛﻪ ﺭﺍﺑﻄﺔ ) (١٢-٣ﺗﻌﺪﺍﺩ ﺑﻴﺖ ﻫﺎ ) (Bﺭﺍ ﺑﺮﺍﺑﺮ ۷ﻗﺮﺍﺭ ﻣﻲ ﺩﻫﺪ ﺗﺎ ) SNRﺣﺪﻭﺩ (36dBﻛﻴﻔﻴﺖ ﻗﺎﺑﻞ ﻗﺒﻮﻟﻲ ﺭﺍ ﺩﺭ ﺍﻏﻠﺐ ﺳﻴﺴﺘﻤﻬﺎﻱ ﻣﺨﺎﺑﺮﺍﺗﻲ ﺗﺄﻣﻴﻦ ﻛﻨﺪ ،ﺑﻪ ﻃﻮﺭ ﻣﻌﻤﻮﻝ ﺗﻌﺪﺍﺩ ﺑﻴﺖ ﻫﺎﻱ ﻣﻮﺭﺩﻧﻴﺎﺯ ﺑﺮﺍﻱ ﺗﺄﻣﻴﻦ ﻛﻴﻔﻴﺖ ﺑﺎﻻﻱ ﺳﻴﮕﻨﺎﻝ ﮔﻔﺘﺎﺭ ﺑﺎ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ١١ ،ﺑﻴﺖ ﺍﺳﺖ. m - law -2-1-2 ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﺮﺩﻥ ﻳﻜﻨﻮﺍﺧﺖ ﺗﻨﻬﺎ ﺑﺮﺍﻱ ﺳﻴﮕﻨﺎﻟﻬﺎﻱ ﺑﺎ ﺗﻮﺯﻳﻊ ﻳﻜﻨﻮﺍﺧﺖ ﺑﻬﻴﻨﻪ ﺍﺳﺖ .ﺑﺮﺍﻱ ﺳﻴﮕﻨﺎﻟﻬﺎﻳﻲ ﻛﻪ ﻧﺰﺩﻳﻚ ﻣﻘﺎﺩﻳﺮ ﻛﻮﭼﻚ ﺩﺍﻣﻨﻪ ﺗﺠﻤﻊ ﺩﺍﺭﻧﺪ ،ﺑﻪ ﻋﻨﻮﺍﻥ ﻣﺜﺎﻝ ﺗﻮﺯﻳﻊ ﮔﻮﺳﻲ ﺑﺎ ﻣﻴﺎﻧﮕﻴﻦ ﺻﻔﺮ ،ﺑﻬﺘﺮ ﺍﺳﺖ ﻛﻪ ﺩﺍﻣﻨﻪ ﻫﺎﻱ ﻛﻮﭼﻚ ﺑﺎ ﺩﻗﺖ ﺑﻴﺸﺘﺮﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﻮﻧﺪ .ﺑﺮﺍﻱ ﺗﺤﻘﻖ ﺍﻳﻦ ﺍﻣﺮ ﺍﺑﺘﺪﺍ ﺑﺎﻳﺪ ﻧﮕﺎﺷﺘﻲ ﺑﻪ ﺳﻴﮕﻨﺎﻝ ﻛﺮﺩ ﺑﻪ ﻃﻮﺭﻱ ﻛﻪ ﻣﻘﺎﺩﻳﺮ ﻛﻮﭼﻚ ﺭﺍ ﺗﻘﻮﻳﺖ ﻛﻨﺪ ﻭ ﺳﭙﺲ ﻳﻚ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﺓ ﻳﻜﻨﻮﺍﺧﺖ ﺑﻪ ﺳﻴﮕﻨﺎﻝ ﻧﮕﺎﺷﺖ ﺷﺪﻩ ﺍﻋﻤﺎﻝ ﻛﺮﺩ .ﻳﻜﻲ ﺍﺯ ﻧﮕﺎﺷﺖﻫﺎ ﺑﻪ ﺻﻮﺭﺕ ﺯﻳﺮ ﺍﺳﺖ. )(١٣-٣ 3 ]). sin [x(n ù ú úû ])y(n) = F [x(n é )x(n log ê1 + m X max êë ] log[1 + m = X max CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ﺷﮑﻞ -2راﺑﻄﻪ ورودي – ﺧﺮوﺟﯽ ﺑﺮاي ﯾﮏ ﻣﺸﺨﺼﻪ ) m - lawاﻗﺘﺒﺎس از ](smith[2 ﺷﮑﻞ ،٢ﻳﻚ ﺧﺎﻧﻮﺍﺩﻩ ﺍﺯ ﻣﻨﺤﻨﻲ ﻫﺎﻱ ) y(nﺑﺮ ﺣﺴﺐ ) x(nﺭﺍ ﺑﺮﺍﻱ ﻣﻘﺎﺩﻳﺮ ﻣﺘﻔﺎﻭﺕ mﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ .ﻭﺍﺿﺢ ﺍﺳﺖ ﻛﻪ ﺑﺎ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﺗﺎﺑﻊ ) (١٣-٣ﺩﺍﻣﻨﻪ ﻫﺎﻱ ﻭﺭﻭﺩﻱ ﻛﻮﭼﻚ ﺗﻘﻮﻳﺖ ﻣﻲ ﺷﻮﻧﺪ .ﺷﮑﻞ ٣ﺗﻮﺯﻳﻊ ﺳﻄﻮﺡ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﺭﺍ ﺑﺮﺍﻱ ﺣﺎﻟﺘﻲ ﻛﻪ m =٤٠ﻭ N=٨ﺍﺳﺖ، ﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ .ﺍﮔﺮ m =٠ﺑﺎﺷﺪ ،ﻣﻌﺎﺩﻟﺔ ) (١٣-٣ﺑﻪ ﻣﻌﺎﺩﻟﺔ ) y(n)=x(nﺧﻼﺻﻪ ﻣﻲ ﺷﻮﺩ ،ﺳﻄﻮﺡ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﺑﺎ ﻓﺎﺻﻠﻪﻫﺎﻱ ﻳﻜﻨﻮﺍﺧﺖ ﺗﻘﺴﻴﻢ ﺷﺪﻩ ﺍﻧﺪ ،ﺑﺎ ﺍﻳﻦ ﻭﺟﻮﺩ ﺑﺮﻱ ﻣﻘﺎﺩﻳﺮ ﺑﺰﺭﮒ mﻭ ﺑﺮﺍﻱ |) |x(nﻫﺎﻱ ﺑﺰﺭﮒ: )(١٤-٣ ﺑﻨﺎﺑﺮﺍﻳﻦ ﺑﻪ ﺟﺰ ﺩﺍﻣﻨﻪ ﻫﺎﻱ ﺑﺴﻴﺎﺭ ﻛﻮﭼﻚ ،ﺳﻄﻮﺡ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﺑﻪ ﻃﻮﺭ ﻧﻤﺎﻳﻲ ﺑﺎ ﺍﻧﺪﻳﺲ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﺍﻓﺰﺍﻳﺶ ﻣﻲ ﻳﺎﺑﻨﺪ .ﺍﻳﻦ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ، ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ m - lawﻧﺎﻣﻴﺪﻩ ﻣﻲ ﺷﻮﺩ ﻭ ﺍﻭﻟﻴﻦ ﺑﺎﺭ ﺗﻮﺳﻂ smithﺍﺭﺍﻳﻪ ﺷﺪ ].[2 ﺷﮑﻞ -3ﺗﻮزﯾﻊ ﺳﻄﻮح ﮐﻮاﻧﺘﯿﺰاﺳﯿﻮن ﺑﺮاي ﮐﻮاﻧﺘﯿﺰه ﮐﻨﻨﺪة 3ﺑﯿﺘﯽ -law mﺑﺎ m =40از ][1 ﺑﺎ ﺑﻜﺎﺭﮔﻴﺮﻱ ﻫﻤﺎﻥ ﻓﺮﺿﻴﺎﺕ ﻛﻪ ﺑﺮﺍﻱ ﺁﻧﺎﻟﻴﺰ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ﺍﺳﺘﻔﺎﺩﻩ ﺷﺪ smith[2] ،ﺭﺍﺑﻄﻪ ﺯﻳﺮ ﺑﺮﺍﻱ ﻧﺴﺒﺖ ﺳﻴﮕﻨﺎﻝ ﺑﻬﻨﻮﻳﺰ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺩﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﺓ m - lawﺑﺪﺳﺖ ﺁﻭﺭﺩ. 4 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 )(١٥-٣ x max ﺍﻳﻦ ﻣﻌﺎﺩﻟﻪ ﺑﺴﺘﮕﻲ ﻛﻤﺘﺮ SNRﺑﻪ ﻣﻘﺪﺍﺭ ) sx ( ﺭﺍ ﻛﻪ ﺑﻪ ﺗﻮﻳﻊ ﺳﻴﮕﻨﺎﻝ ﺑﺴﺘﮕﻲ ﺩﺍﺭﺩ ﺭﺍ ﺩﺭ ﻣﻘﺎﻳﺴﻪ ﺑﺎ ﻣﻌﺎﺩﻟﻪ ) (١٢-٣ﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ. x max ﻣﺸﺎﻫﺪﻩ ﻣﻲﺷﻮﺩ ﻛﻪ ﺑﺎ ﺍﻓﺰﺍﻳﺶ SNR ، mﺑﻪ ﺗﻐﻴﻴﺮﺍﺕ ) sx ( ﻛﻤﺘﺮ ﺑﺴﺘﮕﻲ ﭘﻴﺪﺍ ﻣﻲ ﻛﻨﺪ ،ﻳﻌﻨﻲ ﺑﺎ ﻭﺟﻮﺩ ﺍﻳﻨﻜﻪ ﺗﺮﻡ x max ]) SNR ، - 20log10[ln(1 + mﺭﺍ ﻛﺎﻫﺶ ﻣﻲ ﺩﻫﺪ ،ﻣﺤﺪﻭﺩﻩ ﺍﻱ ﺍﺯ ) sx ﺑﻨﺎﺑﺮﺍﻳﻦ ﺑﺎ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﻳﻚ mﺑﺰﺭﮒ ،ﻛﺎﺭﺁﻳﻲ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﺑﻪ ﺁﻣﺎﺭﮔﺎﻥ ﺳﻴﮕﻨﺎﻝ ﻭﺍﺑﺴﺘﮕﻲ ﻛﻤﺘﺮﻱ ﭘﻴﺪﺍ ﻣﻲ ﻛﻨﺪ. ( ﻛﻪ ﺩﺭ ﺁﻥ SNRﺛﺎﺑﺖ ﺍﺳﺖ ﺑﺎ mﺍﻓﺰﺍﻳﺶ ﻣﻲ ﻳﺎﺑﺪ. -2-2ﮐﺪ ﮐﺮدن ﭘﯿﺸﮕﻮﯾﺎﻧﻪ )(Predictive Coding ﺩﺭ ﻳﻚ ﺷﻜﻞ ﻣﻮﺝ ﺻﻮﺕ ﻣﻌﻤﻮﻟﻲ ،ﻧﻤﻮﻧﻪ ﻫﺎﻱ ﻣﺘﻮﺍﻟﻲ ﺑﺠﺰ ﺩﺭ ﮔﺬﺍﺭﻫﺎﻱ ﺑﻴﻦ ﺁﻭﺍﻫﺎﻱ ﻣﺘﻔﺎﻭﺕ ،ﻣﻘﺎﺩﻳﺮ ﻣﺸﺎﺑﻬﻲ ﺩﺍﺭﻧﺪ .ﻳﻚ ﺭﺍﻩ ﺑﺮﺍﻱ ﺑﻬﺮﻩ ﮔﻴﺮﻱ ﺍﺯ ﺍﻳﻦ ﻫﻤﺒﺴﺘﮕﻲ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﻛﺪ ﻛﺮﺩﻥ ﺑﻪ ﺭﻭﺵ ﭘﻴﺸﮕﻮﻳﻲ ﺧﻄﻲ ﺍﺳﺖ .ﺍﺑﺘﺪﺍ ﻧﻤﻮﻧﻪ ﻓﻌﻠﻲ ) x(nﺍﺯ ﺭﻭﻱ ﺗﺮﻛﻴﺐ ﺧﻄﻲ ﻧﻤﻮﻧﻪ ﻫﺎﻱ ﻗﺒﻠﻲ ﺳﺎﺧﺘﻪ ﺷﺪﻩ ) x(n - kﺗﺨﻤﻴﻦ ﺯﺩﻩ ﻣﻲ ﺷﻮﺩ ﺗﺎ ﺳﭙﺲ ﺧﻄﺎﻱ ﺑﻴﻦ ﻣﻘﺪﺍﺭ ﻧﻤﻮﻧﻪ ﺍﺻﻠﻲ ﻭ ﻣﻘﺪﺍﺭ ﭘﻴﺶ ﺑﻴﻨﻲ ﺷﺪﻩ ﺑﻪ ) d(nﻛﻮﺍﻧﺘﻴﺰﻩ ﻣﻲ ﺷﻮﺩ ﻭ ﺑﻮﺳﻴﻠﺔ ﻛﻠﻤﺔ ﻛﺪ ) ،c(nﻛﺪ ﻣﻲ ﺷﻮﺩ. ﺩﺭ ﺩﻱ ﻛﺪ ﻛﻨﻨﺪﻩ ،ﺍﺑﺘﺪﺍ ﻫﻤﺎﻥ ﻣﻘﺪﺍﺭ ﭘﻴﺸﮕﻮﻳﻲ ﺷﺪﻩ ﺍﺯ ﺭﻭﻱ ﻧﻤﻮﻧﻪ ﻫﺎﻱ ﻗﺒﻠﻲ ﺩﻱ ﻛﺪ ﺷﺪﻩ ﺳﺎﺧﺘﻪ ﻣﻲ ﺷﻮﺩ .ﺍﻳﻦ ﻣﻘﺪﺍﺭ ﺳﭙﺲ ﺑﻪ ﻣﻘﺪﺍﺭ ﺧﻄﺎﻱ ﺩﻱ ﻛﺪ ﻭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺍﺿﺎﻓﻪ ﻣﻲ ﺷﻮﺩ ﺗﺎ ﻣﻘﺪﺍﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺑﺮﺍﻱ ﻧﻤﻮﻧﻪ ﻓﻌﻠﻲ ﺑﺪﺳﺖ ﺁﻳﺪ .ﻳﻌﻨﻲ: ﺑﻠﻮﻙ ﺩﻳﺎﮔﺮﺍﻡ ﻛﺪ ﻛﻨﻨﺪﻩ ﻭ ﺩﻱ ﻛﺪ ﻛﻨﻨﺪﻩ ﻳﻚ ﺳﻴﺴﺘﻢ ﻛﺪ ﻛﻨﻨﺪﺓ ﭘﻴﺸﮕﻮ ﺩﺭ ﺷﮑﻞ ٤ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﺷﺪﻩ ﺍﺳﺖ .ﺳﻴﺴﺘﻢ ﻛﺪ ﻛﻨﻨﺪﻩ ﭘﻴﺸﮕﻮ ﻣﻌﻤﻮﻵً ﺑﻪ ﻣﺪﻭﻻﺳﻴﻮﻥ ﻛﺪ ﺷﺪﻩ ﺳﻴﮕﻨﺎﻝ ﺗﻔﺎﺿﻠﻲ ﻳﺎ ” “DPCMﺷﻨﺎﺧﺘﻪ ﻣﻲ ﺷﻮﺩ .ﻛﻠﻤﺔ »ﺗﻔﺎﺿﻠﻲ« ﺑﻪ ﺍﻳﻦ ﻣﻮﺿﻮﻉ ﺍﺷﺎﺭﻩ ﻣﻲ ﻛﻨﺪ ﻛﻪ ﺳﻴﮕﻨﺎﻝ ﺧﻄﺎﻱ ﭘﻴﺸﮕﻮﻳﻲ ﻛﺪ ﻣﻲﺷﻮﺩ ﻭ ” “PCMﺑﻪ ﻃﺮﺡ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺍﺷﺎﺭﻩ ﻣﻲ ﻛﻨﺪ ﻛﻪ ﺩﺭ ﺁﻥ ﻫﺮ ﺑﻴﺖ ﻛﺪ ﺷﺪﻩ ﻳﻚ ﺳﻤﺒﻞ ﺍﺳﺖ ﻛﻪ ﺑﻮﺳﻴﻠﻪ ﻳﻚ ﭘﺎﻟﺲ )ﺑﺎ ﺩﺍﻣﻨﻪ ﺻﻔﺮ ﻳﺎ ﻳﻚ( ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﻣﻲ ﺷﻮﺩ .ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﺮﺩﻥ ﻣﺴﺘﻘﻴﻢ ﻳﻚ ﻧﻤﻮﻧﺔ ﺍﻭﻟﻴﻪ ﺑﺎ ﻃﻮﻝ ﺛﺎﺑﺖ ﺩﺭ ﻛﺪ ﻛﺮﺩﻥ “PCM” ،ﻧﺎﻣﻴﺪﻩ ﻣﻲ ﺷﻮﺩ. 5 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ﺷﮑﻞ -4ﮐﺪ ﮐﺮدن ﭘﯿﺸﮕﻮﯾﺎﻧﻪ )اﻟﻒ( ﮐﺪ ﮐﻨﻨﺪه )ب( دي ﮐﺪ ﮐﻨﻨﺪه -1-2-2ﻣﺪوﻻﺳﯿﻮن دﻟﺘﺎ ﻳﻚ ﺳﻴﺴﺘﻢ ﺳﺎﺩﺓ ﭘﻴﺸﮕﻮﻳﺎﻧﻪ ،ﺳﻴﺴﺘﻢ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ) (DMﺍﺳﺖ ﻛﻪ ﺩﺭ ﺷﮑﻞ ٥ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﺷﺪﻩ ﺍﺳﺖ .ﺩﺭ ﺍﻳﻦ ﺳﻴﺴﺘﻢ ،ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﺓ ﺧﻄﺎﻱ ﭘﻴﺸﮕﻮﻳﻲ ﻓﻘﻂ ﺩﻭ ﺳﻄﺢ ﺩﺍﺭﺩ ﻭ ﻃﻮﻝ ﭘﻠﻪ »ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ« ﺛﺎﺑﺖ ﺍﺳﺖ .ﺳﻄﺢ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻣﺜﺒﺖ ﺑﺎ c(n)=0ﻭ ﻣﻨﻔﻲ ﺑﺎ c(n)=1 ﻣﺸﺨﺺ ﻣﻲ ﺷﻮﺩ .ﺑﻨﺎﺑﺮﺍﻳﻦ ) d(nﺑﻪ ﺻﻮﺭﺕ ﺯﻳﺮ ﺗﻌﺮﻳﻒ ﻣﻲ ﺷﻮﺩ. )(١٦-٣ ﻛﻪ ﺍﺯ ﻳﻚ ﭘﻴﺸﮕﻮﻳﻲ ﺧﻄﻲ ﺩﺭﺟﻪ ﺍﻭﻝ ﺍﺳﺘﻔﺎﺩﻩ ﺷﺪﻩ ﺍﺳﺖ ،ﻳﻌﻨﻲ ) xp(n) = x(n-1ﻣﻲ ﺗﻮﺍﻥ ﺍﺯ ﺷﮑﻞ-٥ﺍﻟﻒ ﻣﺸﺎﻫﺪﻩ ﻛﺮﺩ ﻛﻪ ﻣﻌﻤﻮﻻﹰ ) x(nﺩﺭ ﻣﻌﺎﺩﻟﻪ ﺗﻔﺎﺿﻠﻲ ﺯﻳﺮ ﺻﺪﻕ ﻣﻲ ﻛﻨﺪ. )(١٧-٣ ﺑﺎ ، a = 1ﺍﻳﻦ ﻣﻌﺎﺩﻟﻪ ،ﻣﻌﺎﺩﻝ ﺩﻳﺠﻴﺘﺎﻟﻲ ﺍﻧﺘﮕﺮﺍﻝ ﺍﺳﺖ .ﻫﻤﭽﻨﻴﻦ ﺑﺎﻳﺪ ﺩﻗﺖ ﻛﺮﺩ ﻛﻪ ﻭﺭﻭﺩﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ )(١٨-٣ ﻣﻲ ﺑﺎﺷﺪ .ﺑﻨﺎﺑﺮﺍﻳﻦ ﺑﻪ ﺟﺰ ﺧﻄﺎﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺩﺭ ) d(n) ، x(n - 1ﻳﻚ ﺗﻔﺎﺿﻠﻲ ﺑﺮﮔﺸﺘﻲ ﺩﺭﺟﻪ ﺍﻭﻝ ﺍﺯ ) x(nﺍﺳﺖ ﻛﻪ ﻣﻲ ﺗﻮﺍﻧﺪ ﺑﻪ ﻋﻨﻮﺍﻥ ﺗﻘﺮﻳﺐ ﺩﻳﺠﻴﺘﺎﻟﻲ ﺑﺮﺍﻱ ﻣﺸﺘﻘﺎﺕ ﻭﺭﻭﺩﻱ ﺩﺭ ﻧﻈﺮ ﮔﺮﻓﺘﻪ ﺷﻮﺩ ،ﻣﻌﻜﻮﺱ ﭘﺮﻭﺳﺔ ﺍﻧﺘﮕﺮﺍﻝ ﺩﻳﺠﻴﺘﺎﻟﻲ. ﺍﺯ ﺁﻧﺠﺎ ﻛﻪ ﺧﻄﺎﻱ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻓﻘﻂ ﺩﻭ ﺳﻄﺢ ﺩﺍﺭﺩ ،ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ﺩﺍﺭﺍﻱ ﻧﺮﺥ ﺑﻴﺘﻲ ﺑﺮﺍﺑﺮ ۱bit/sampleﺍﺳﺖ .ﺍﮔﺮ ﺑﻪ ﺩﻧﺒﺎﻟﻪ ١٦bits/sampleﺍﻋﻤﺎﻝ ﺷﻮﺩ ،ﺁﻧﮕﺎﻩ ﺑﻪ ﻧﺮﺥ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ) (CRﺑﺮﺍﺑﺮ ١٦ﻣﻲ ﺭﺳﺪ. 6 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ﺷﮑﻞ -5ﺑﻠﻮك دﯾﺎﮔﺮام ﺳﯿﺴﺘﻢ ﻣﺪوﻻﺳﯿﻮن دﻟﺘﺎ اﻟﻒ( ﮐﺪ ﮐﻨﻨﺪه ب( دي ﮐﺪ ﮐﻨﻨﺪه ﺑﺮﺍﻱ ﺍﻳﻨﻜﻪ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ﺑﻪ ﺧﻮﺑﻲ ﻛﺎﺭ ﻛﻨﺪ ،ﺍﻧﺪﺍﺯﻩ ﭘﻠﻪ ﺑﺎﻳﺪ ﻃﻮﺭﻱ ﺍﻧﺘﺨﺎﺏ ﺷﻮﺩ ﻛﻪ ﺗﻐﻴﻴﺮﺍﺕ ﺳﻴﮕﻨﺎﻝ ﺭﺍ ﺩﻧﺒﺎﻝ ﻛﻨﺪ .ﺗﺤﻘﻖ ﺍﻳﻦ ﺍﻣﺮ ﻣﺸﻜﻞ ﺍﺳﺖ ﺯﻳﺮﺍ ﻣﺸﺨﺼﺎﺕ ﺳﻴﮕﻨﺎﻝ ﺍﺯ ﻳﻚ toneﺑﻪ toneﺩﻳﮕﺮ ﺗﻐﻴﻴﺮ ﻣﻲ ﻛﻨﺪ .ﺷﮑﻞ-٦ﺍﻟﻒ ﭘﺮﻭﺳﻪ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ﺭﺍ ﺑﺎ ﻃﻮﻝ ﭘﻠﻪ ﻣﺘﻨﺎﺳﺐ ﻭ ﺩﻗﻴﻖ ﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ .ﻣﻲ ﺗﻮﺍﻥ ﻣﺸﺎﻫﺪﻩ ﻛﺮﺩ ﻛﻪ ﻃﻮﻝ ﭘﻠﻪ ﺩﺭ ﺍﺑﺘﺪﺍ ﺑﺴﻴﺎﺭ ﻛﻮﭼﻚ ﺍﺳﺖ ﻛﻪ ﺑﺎﻋﺚ ﻣﻲﺷﻮﺩ ﺳﻴﮕﻨﺎﻝ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺯﻳﺮ ﺩﺍﻣﻨﻪ ﺳﻴﮕﻨﺎﻝ ﺍﺻﻠﻲ ﺁﻫﺴﺘﻪ ﺗﺮ ﺣﺮﻛﺖ ﻛﻨﺪ .ﺍﺯ ﻃﺮﻑ ﺩﻳﮕﺮ ﺍﮔﺮ ﻃﻮﻝ ﭘﻠﻪ ﺭﺍ ﺧﻴﻠﻲ ﺑﺰﺭﮒ ﺑﮕﻴﺮﻳﻢ ،ﺑﺎﻋﺚ ﻣﻲ ﺷﻮﺩ ﻛﻪ ﺳﻴﮕﻨﺎﻝ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺩﺭ ﺣﻮﻝ ﻭ ﺣﻮﺵ ﺳﻴﮕﻨﺎﻝ ﺍﺻﻠﻲ ﻧﻮﺳﺎﻥ ﻛﻨﺪ .ﺑﺮﺍﻱ ﻛﺎﺭﺁﻳﻲ ﺑﻬﺘﺮ ،ﻃﻮﻝ ﭘﻠﻪ ﺑﺎﻳﺪ ﺑﻪ ﻃﻮﺭ ﻭﻓﻘﻲ ﺑﺎﺷﺪ ﻛﻪ ﻣﻮﺿﻮﻉ ﺑﺨﺶ ﺁﻳﻨﺪﻩ ﺍﺳﺖ. 7 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ﺷﮑﻞ-6ﻧﻤﺎﯾﺶ ﻣﺪوﻻﺳﯿﻮن دﻟﺘﺎ اﻟﻒ( اﺳﺘﻔﺎده از ﯾﮏ ﭘﻠﻪ ﺑﺎ ﻃﻮل ﺛﺎﺑﺖ ب( اﺳﺘﻔﺎده از ﻃﻮل ﭘﻠﻪ وﻓﻘﯽ -2-2-2ﻣﺪوﻻﺳﯿﻮن دﻟﺘﺎي وﻓﻘﯽ ﻃﺮﺣﻬﺎﻱ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎﻱ ﻭﻓﻘﻲ ) (ADMﻣﺘﻌﺪﺩﻱ ﭘﻴﺸﻨﻬﺎﺩ ﺷﺪﻩ ﺍﻧﺪ .ﺑﻴﺸﺘﺮ ﺍﻳﻦ ﻃﺮﺣﻬﺎ ﺍﺯ ﻧﻮﻉ ﺑﺮﮔﺸﺘﻲ ﻫﺴﺘﻨﺪ ﻛﻪ ﺩﺭ ﺁﻧﻬﺎ ﻃﻮﻝ ﭘﻠﻪ ﺑﺮﺍﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﺓ ﺩﻭ ﺳﻄﺤﻲ ﺑﺮ ﻣﺒﻨﺎﻱ ﻛﻠﻤﺎﺕ ﻛﺪ ﺩﺭ ﺧﺮﻭﺟﻲ ﺑﻬﻴﻨﻪ ﻣﻲ ﺷﻮﺩ .ﺳﻴﺴﺘﻤﻲ ﻛﻪ ﻣﺎ ﺩﺭ ﺯﻳﺮ ﭘﻴﺸﻨﻬﺎﺩ ﻛﺮﺩﻩ ﺍﻳﻢ ﺑﻮﺳﻴﻠﺔ ] Jayant[3ﻃﺮﺍﺣﻲ ﺷﺪﻩ ﺍﺳﺖ .ﻃﻮﻝ ﭘﻠﻪ ﺩﺭ ﺍﻟﮕﻮﺭﻳﺘﻢ Jayantﺍﺯ ﻗﺎﻧﻮﻥ ﺯﻳﺮ ﭘﻴﺮﻭﻱ ﻣﻲ ﻛﻨﺪ. )-١٩-٣ﺍﻟﻒ( )-١٩-٣ﺏ( ﺍﻟﮕﻮﺭﻳﺘﻢ ﺑﺮﺍﻱ ﺗﻌﻴﻴﻦ ﻃﻮﻝ ﭘﻠﻪ ﺑﻪ ﺻﻮﺭﺕ ﺯﻳﺮ ﺍﺳﺖ. )(٢٠-٣ ﺷﮑﻞ-٦ﺏ -ﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ ﻛﻪ ﭼﮕﻮﻧﻪ ﺷﻜﻞ ﻣﻮﺝ ﺷﮑﻞ-٦ﺍﻟﻒ ﻣﻲ ﺗﻮﺍﻧﺪ ﺗﻮﺳﻂ ﻳﻚ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎﻱ ﻭﻓﻘﻲ ﻛﻪ ﺩﺭ ﺭﺍﺑﻄﻪ ) (١٨-٣ﻭ ) (٢٠-٣ﺑﻴﺎﻥ ﺷﺪ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﻮﺩ .ﺑﺮﺍﻱ ﺭﺍﺣﺘﻲ ،ﭘﺎﺭﺍﻣﺘﺮﻫﺎﻱ ﺳﻴﺴﺘﻢ ﺩﺭ p = 2ﻭ a = 1ﺗﻨﻈﻴﻢ ﻣﻲ ﺷﻮﻧﺪ ﻭ ﺣﺪﺍﻗﻞ ﻃﻮﻝ ﭘﻠﻪ ﺩﺭ ﺷﻜﻞ ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﺷﺪﻩ ﺍﺳﺖ .ﻣﻲ ﺗﻮﺍﻥ ﻣﺸﺎﻫﺪﻩ ﻛﺮﺩ ﻛﻪ ﻧﻮﺍﺣﻲ ﺑﺎ ﺷﻴﺐ ﻣﺜﺒﺖ ﺯﻳﺎﺩ ﻫﻨﻮﺯ ﻳﻚ ﺩﻧﺒﺎﻟﻪ ﺍﺯ ﺻﻔﺮ ﺗﻮﻟﻴﺪ ﻣﻲ ﻛﻨﻨﺪ ﺍﻣﺎ ﺩﺭ ﺍﻳﻦ ﺣﺎﻟﺖ ﻃﻮﻝ ﭘﻠﻪ ﺁﻧﻘﺪﺭ ﺍﻓﺰﺍﻳﺶ ﻣﻲ ﻳﺎﺑﺪ ﺗﺎ ﺍﺯﺩﻳﺎﺩ ﺷﻴﺐ ﺷﻜﻞ ﻣﻮﺝ ﺭﺍ ﺩﻧﺒﺎﻝ ﻛﻨﺪ .ﻧﻮﺍﺣﻲ ﺩﺍﻧﻪ ﺩﺍﻧﻪ ﺍﻱ ﺩﺭ ﺳﻤﺖ ﺭﺍﺳﺖ ﺷﻜﻞ ﺩﻭﺑﺎﺭﻩ ﺑﻮﺳﻴﻠﻪ ﻳﻚ ﺩﻧﺒﺎﻟﻪ 8 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ﺍﺯ ﺻﻔﺮ ﻭ ﻳﻚ ﻫﺎﻱ ﻣﺘﻨﺎﺳﺐ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻣﻲﺷﻮﻧﺪ ،ﺍﻣﺎ ﺩﺭ ﺍﻳﻦ ﺣﺎﻟﺖ ﻃﻮﻝ ﭘﻠﻪ ﺳﺮﻳﻌﺎﹰ ﺑﻪ ﻣﻘﺪﺍﺭ ﺣﺪﺍﻗﻞ ) (D minﻛﺎﻫﺶ ﻣﻲ ﻳﺎﺑﺪ ﻭ ﺗﺎ ﻭﻗﺘﻲ ﻛﻪ ﺷﻴﺐ ﻛﻢ ﺑﺎﺷﺪ ﺩﺭ ﺍﻳﻦ ﻣﻘﺪﺍﺭ ﻣﻲ ﻣﺎﻧﺪ. ﺷﮑﻞ -7ﻧﺴﺒﺖ ﻫﺎي ﺳﯿﮕﻨﺎل ﺑﻪ ﻧﻮﯾﺰ از ﯾﮏ ﻣﺪوﻻﺗﻮر دﻟﺘﺎي وﻓﻘﯽ ﺑﺮ ﺣﺴﺐ ﺗﻮاﺑﻊ p ﺷﮑﻞ ،٧ﻧﺘﺎﻳﺞ ﺷﺒﻴﻪ ﺳﺎﺯﻱ ﺭﺍ ﺑﺮﺍﻱ ﺳﻴﮕﻨﺎﻝ ﮔﻔﺘﺎﺭ ﺑﺎ PQ = 1ﺑﺮﺍﻱ ﺳﻪ ﻧﺮﺥ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﻣﺘﻔﺎﻭﺕ ﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ. ﻣﺸﺎﻫﺪﻩ ﻣﻲ ﺷﻮﺩ ﻛﻪ ﺣﺪﺍﻛﺜﺮ SNRﺑﺮﺍﻱ P = ١/٥ﺑﺪﺳﺖ ﻣﻲ ﺁﻳﺪ ،ﺑﺎ ﺍﻳﻦ ﻭﺟﻮﺩ ،ﻗﻠﺔ ﻣﻨﺤﻨﻲ ﺑﺴﻴﺎﺭ ﭘﻬﻦ ﺍﺳﺖ ﻭ SNRﭼﻨﺪ dBﺑﺎﻻﺗﺮ ﻭ ﭘﺎﻳﻴﻦ ﺗﺮ ﺍﺯ ﻣﻘﺪﺍﺭ ﺣﺪﺍﻛﺜﺮ ﺑﺮﺍﻱ ١/٢٥<p<٢ﻗﺮﺍﺭ ﺩﺍﺭﺩ .ﺗﻮﺟﻪ ﻛﻨﻴﺪ ﺑﺮﺍﻱ ﺍﻳﻨﻜﻪ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ﺧﻮﺏ ﻛﺎﺭ ﻛﻨﺪ ﺑﺎﻳﺪ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺍﺯ ﺳﻴﮕﻨﺎﻝ ﺩﺭ ﻓﺮﻛﺎﻧﺴﻲ ﺑﺎﻻﺗﺮ ﺍﺯ ﺁﻧﭽﻪ ﺗﺌﻮﺭﻱ ﻧﺎﻳﻜﻮﺋﻴﺴﺖ ﺗﺤﻤﻴﻞ ﻣﻲ ﻛﻨﺪ ،ﺍﻧﺠﺎﻡ ﺷﻮﺩ ﺗﺎ ﺗﻐﻴﻴﺮﺍﺕ ﺑﻴﻦ ﻧﻤﻮﻧﻪ ﻫﺎﻱ ﻣﺘﻮﺍﻟﻲ ﻛﻮﭼﻚ ﺑﺎﺷﺪ .ﺍﻳﻦ ﭘﺪﻳﺪﻩ ﺩﺭ ﺣﻘﻴﻘﺖ ﻣﺼﺎﻟﺤﻪ ﺑﻴﻦ ﺩﻗﺖ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﻭ ﺩﻗﺖ ﺩﺍﻣﻨﻪ ﺭﺍ ﺁﺷﻜﺎﺭ ﻣﻲ ﻛﻨﺪ .ﻳﻌﻨﻲ ﺑﺮﺍﻱ ﻛﺎﻫﺶ ﺩﻗﺖ ﺩﺍﻣﻨﻪ )ﻳﻚ ﺑﻴﺖ ﺩﺭ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ( ﺑﺎﻳﺪ ﺩﻗﺖ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺭﺍ ﺍﻓﺰﺍﻳﺶ ﺩﺍﺩ. ﺩﺭ ﺍﻳﻦ ﺁﺯﻣﺎﻳﺶ ﺷﻤﺎ ﺑﺎ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ﺩﺭ ﺣﺎﻟﺖ ﻃﻮﻝ ﭘﻠﻪ ﺛﺎﺑﺖ ﻭ ﻭﻓﻘﻲ ﻛﺎﺭ ﺧﻮﺍﻫﻴﺪ ﻛﺮﺩ. DPCM -3-2-2ﻫﺎي ﻣﺮﺗﺒﻪ ﺑﺎﻻﺗﺮ ﻣﺪﻭﻻﺗﻮﺭﻫﺎﻱ ﺩﻟﺘﺎ ،ﻫﻤﺎﻧﻄﻮﺭ ﻛﻪ ﺩﺭ ﺑﺨﺶ ﻗﺒﻠﻲ ﺑﻴﺎﻥ ﺷﺪ ،ﻣﻲ ﺗﻮﺍﻧﻨﺪ ﺳﻴﺴﺘﻤﻬﺎﻱ DPCMﻳﻚ ﺑﻴﺘﻲ ﻧﺎﻣﻴﺪﻩ ﺷﻮﻧﺪ .ﺑﻪ ﻃﻮﺭ ﻛﻠﻲ ،ﻣﻲ ﺗﻮﺍﻥ ﺍﺯ ﺑﻴﺸﺘﺮ ﺍﺯ ﻳﻚ ﻧﻤﻮﻧﻪ ﻗﺒﻠﻲ ﺑﺮﺍﻱ ﺗﺨﻤﻴﻦ ﻧﻤﻮﻧﻪ ﻓﻌﻠﻲ ﺍﺳﺘﻔﺎﺩﻩ ﻛﺮﺩ .ﻫﻤﭽﻨﻴﻦ ،ﻣﻲ ﺗﻮﺍﻥ ﺍﺯ ﻳﻚ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﺑﺎ ﺑﻴﺸﺘﺮ ﺍﺯ ﺩﻭ ﺳﻄﺢ ﺍﺳﺘﻔﺎﺩﻩ ﻛﺮﺩ .ﺑﺮﺍﻱ ﺩﺭﻙ ﺑﻬﺘﺮ ﺍﺯ ﭼﮕﻮﻧﮕﻲ ﺗﻌﻴﻴﻦ ﺿﺮﺍﻳﺐ ﭘﻴﺸﮕﻮ ﻭ ﻃﺮﺍﺣﻲ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﺑﻬﻴﻨﻪ ﺑﻪ ﻣﺮﺟﻊ ] [1ﻣﺮﺍﺟﻌﻪ ﻛﻨﻴﺪ .ﻋﻤﻮﻣﺎﹰDPCM ، ﺑﺮﺍﻱ ﺳﻴﺴﺘﻤﻬﺎﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺗﻔﺎﺿﻠﻲ ﻛﻪ ﺩﺭ ﺁﻥ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﺑﻴﺸﺘﺮ ﺍﺯ ﺩﻭ ﺳﻄﺢ ﺩﺍﺭﺩ ،ﻣﻌﻜﻮﺱ ﻣﻲ ﺷﻮﺩ .ﺳﻴﺴﺘﻤﻬﺎﻱ DPCMﺑﺎ ﭘﻴﺸﮕﻮﻫﺎﻱ ﺛﺎﺑﺖ ﻣﻲ ﺗﻮﺍﻧﻨﺪ ﺍﺯ ٤ﺗﺎ dB ١١ﺑﻬﺒﻮﺩ ﺩﺭ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻣﺴﺘﻘﻴﻢ ) (PCMﺍﻳﺠﺎﺩ ﻛﻨﻨﺪ .ﺑﻴﺸﺘﺮﻳﻦ ﺑﻬﺒﻮﺩ ﺩﺭ ﻣﺮﺣﻠﻪ ﺗﻐﻴﻴﺮ ﺍﺯ ﺣﺎﻟﺖ ﺑﺪﻭﻥ ﭘﻴﺸﮕﻮﻳﻲ ﺑﻪ ﺣﺎﻟﺖ ﭘﻴﺸﮕﻮﻳﻲ ﺩﺭﺟﻪ ﺍﻭﻝ ﺭﺥ ﻣﻲ ﺩﻫﺪ .ﺍﻳﻦ ﺑﻬﺒﻮﺩ ﺩﺭ ﮔﺬﺭ ﺑﻪ ﭘﻴﺸﮕﻮﻫﺎﻱ ﺩﺭﺟﻪ ٤ﻳﺎ ٥ﻛﻤﺘﺮ ﻣﺤﺴﻮﺱ ﻣﻲﺑﺎﺷﺪ .ﺩﺭ ﮔﻔﺘﺎﺭ ﺍﺯ ﭘﻴﺸﮕﻮﻫﺎﻱ ﺑﺎ ﺩﺭﺟﻪ ﺑﺎﻻﺗﺮ ﺍﺯ ١٠ﺍﺳﺘﻔﺎﺩﻩ ﻣﻲ ﺷﻮﺩ ،ﺯﻳﺮﺍ ﺳﻴﮕﻨﺎﻝ ﮔﻔﺘﺎﺭ ﺑﺎ ﺩﺭﺟﺎﺕ ﺑﺎﻻﺗﺮ ﺑﻬﺘﺮ ﻣﻲ ﺗﻮﺍﻧﺪ ﻣﺪﻝ ﺷﻮﺩ. ﺑﻬﺮﻩ SNRﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ ﻛﻪ ﺩﺭ ﻳﻚ ﺳﻴﺴﺘﻢ ،DPCMﺑﺪﺳﺖ ﺁﻭﺭﺩﻥ SNRﺧﻮﺍﺳﺘﻪ ﺷﺪﻩ ﺑﺎ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﺑﻴﺖ ﻫﺎﻱ ﻛﻤﺘﺮ ﺍﺯ ﺑﻴﺖ ﻫﺎﻱ ﻣﻮﺭﺩﻧﻴﺎﺯ ،ﻫﻨﮕﺎﻣﻲ ﻛﻪ ﻫﻤﺎﻥ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﺑﻪ ﻃﻮﺭ ﻣﺴﺘﻘﻴﻢ ﺭﻭﻱ ﺳﻴﮕﻨﺎﻝ ﮔﻔﺘﺎﺭ ﺍﺛﺮ ﻣﻲ ﻛﻨﺪ ،ﻋﻤﻠﻲ ﺍﺳﺖ .ﺑﻪ ﻳﺎﺩﺁﻭﺭﻳﺪ ﻛﻪ ﻫﻨﮕﺎﻡ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﺮﺩﻥ ﻣﺴﺘﻘﻴﻢ ﻳﻚ ﺳﻴﮕﻨﺎﻝ ،ﻫﺮ ﺑﻴﺖ ﺍﺿﺎﻓﻲ ﺑﻪ ﺑﻬﺮﺓ 6dBﻣﻨﺠﺮ ﻣﻲ ﺷﺪ ،ﺑﻨﺎﺑﺮﺍﻳﻦ ﺍﮔﺮ ﺳﻴﺴﺘﻢ DPCMﺑﺘﻮﺍﻧﺪ ﺑﻪ ﺑﻬﺮﻩ ﭘﻴﺸﮕﻮﻳﻲ 6dB ﺑﺮﺳﺪ ﺑﻪ ﺍﻳﻦ ﻣﻌﻨﻲ ﺍﺳﺖ ﻛﻪ ﻳﻚ ﺑﻴﺖ ﻛﻤﺘﺮ ﻧﺴﺒﺖ ﺑﻪ ﺣﺎﻟﺘﻲ ﻛﻪ ﺳﻴﺴﺘﻢ PCMﻭﺟﻮﺩ ﺩﺍﺷﺘﻪ ﺑﺎﺷﺪ ،ﻣﻮﺭﺩﻧﻴﺎﺯ ﺍﺳﺖ ﺗﺎ ﺑﻪ ﻫﻤﺎﻥ ﻛﻴﻔﻴﺖ ﺍﺯ ﺳﻴﮕﻨﺎﻝ ﺑﺮﺳﺪ. 9 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ADPCM -4-2-2 ﺩﻭ ﻃﺮﺡ ﻋﻤﺪﻩ ﺑﺮﺍﻱ DPCMﻭﻓﻘﻲ ﻳﺎ ADPCMﻭﺟﻮﺩ ﺩﺍﺭﺩ .ﻳﻜﻲ ﺍﺯ ﺁﻧﻬﺎ DPCMﺑﺎ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻭﻓﻘﻲ ﻭ ﺩﻳﮕﺮﻱ DPCMﺑﺎ ﭘﻴﺸﮕﻮﻳﻲ ﻭﻓﻘﻲ ﺍﺳﺖ. ﺑﺮﺍﻱ DPCMﺑﺎ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻭﻓﻘﻲ ،ﻃﻮﻝ ﭘﻠﻪ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﻣﺘﻨﺎﺳﺐ ﺑﺎ ﻭﺍﺭﻳﺎﻧﺲ ﻭﺭﻭﺩﻱ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻨﺪﻩ ﺗﻐﻴﻴﺮ ﻣﻲ ﻛﻨﺪ .ﺑﺎ ﺍﻳﻦ ﻭﺟﻮﺩ ،ﺍﺯ ﺁﻧﺠﺎﻳﻲ ﻛﻪ ﺳﻴﮕﻨﺎﻝ ﺗﻔﺎﺿﻞ ) d(nﻣﺘﻨﺎﺳﺐ ﺑﺎ ﻭﺭﻭﺩﻱ ﺍﺳﺖ ،ﻣﻌﻘﻮﻝ ﺍﺳﺖ ﻛﻪ ﺗﻨﻈﻴﻢ ﻃﻮﻝ ﭘﻠﻪ ﺍﺯ ﺭﻭﻱ ﺳﻴﮕﻨﺎﻝ ﻭﺭﻭﺩﻱ ) x(nﺍﻧﺠﺎﻡ ﺷﻮﺩ ﻛﻪ ﺩﺭ ﺷﮑﻞ ٨ﻧﺸﺎﻥ ﺩﺍﺩﻩ ﺷﺪﻩ ﺍﺳﺖ .ﺭﻭﻳﻪ ﻫﺎﻱ ﻭﻓﻘﻲ ﻣﺘﻌﺪﺩﻱ ﺑﺮﺍﻱ ﺗﻨﻈﻴﻢ ﻃﻮﻝ ﭘﻠﻪ ﺩﺭ ﮔﺬﺷﺘﻪ ﺍﺭﺍﺋﻪ ﺷﺪﻩ ﺍﻧﺪ .ﻧﺘﺎﻳﺞ ﻧﺸﺎﻥ ﻣﻲ ﺩﻫﺪ ﻛﻪ ﺍﻳﻦ ﻗﺒﻴﻞ ﺭﻭﻳﻪ ﻫﺎﻱ ﻭﻓﻘﻲ ﻣﻲ ﺗﻮﺍﻧﻨﺪ ﺩﺭ ﺣﺪﻭﺩ ٥ dBﺩﺭ SNRﻧﺴﺒﺖ ﺑﻪ ﺣﺎﻟﺖ ﻏﻴﺮ ﻭﻓﻘﻲ m - lawﺩﺭ PCMﺑﻬﺒﻮﺩ ﺍﻳﺠﺎﺩ ﻛﻨﻨﺪ .ﺍﻳﻦ ﺑﻬﺒﻮﺩ ﻣﻲ ﺗﻮﺍﻧﺪ ﺑﺎ ٦ dBﺑﻬﺒﻮﺩ ﻛﻪ ﺍﺯ ﻭﺿﻌﻴﺖ ﺗﻔﺎﺿﻠﻲ ﺑﺎ ﭘﻴﺸﮕﻮﻳﻲ ﺛﺎﺑﺖ ﺑﺪﺳﺖ ﻣﻲ ﺁﻳﺪ ،ﺗﺮﻛﻴﺐ ﺷﺪﻩ ﻭ ADPCMﺑﺎ ﭘﻴﺸﮕﻮﻳﻲ ﻭﻓﻘﻲ ﺭﻭ ﺑﻪ ﺟﻠﻮ ،ﺑﻬﺒﻮﺩ SNRﺍﻱ ﺑﺮﺍﺑﺮ 10-11dBﻧﺴﺒﺖ ﺑﻪ PCMﺑﺎ ﻫﻤﺎﻥ ﺗﻌﺪﺍﺩ ﺳﻄﻮﺡ ،ﻧﺘﻴﺠﻪ ﺩﻫﺪ. ﺷﮑﻞ -8ﺳﯿﺴﺘﻢ ADPCMﺑﺎ ﮐﻮاﻧﺘﯿﺰه وﻓﻘﯽ رو ﺑﻪ ﺟﻠﻮ اﻟﻒ( ﮐﺪ ﮐﻨﻨﺪه ب( دي ﮐﺪ ﮐﻨﻨﺪه ﺑﺮﺍﻱ DPCMﺑﺎ ﭘﻴﺸﮕﻮﻳﻲ ﻭﻓﻘﻲ ،ﺿﺮﺍﻳﺐ ﭘﻴﺸﮕﻮﻳﻲ ﻛﻨﻨﺪﻩ ﺑﺴﺘﮕﻲ ﺑﻪ ﺯﻣﺎﻥ ﺩﺍﺭﻧﺪ ،ﺑﻨﺎﺑﺮﺍﻳﻦ ﻣﻘﺎﺩﻳﺮ ﭘﻴﺸﮕﻮﻳﻲ ﺷﺪﻩ ﺑﻪ ﺻﻮﺭﺕ ﺯﻳﺮ ﻫﺴﺘﻨﺪ. )(٢١-٣ ﺩﺭ ﺗﻄﺒﻴﻖ ﺿﺮﺍﻳﺐ ﭘﻴﺸﮕﻮﻳﻲ ) ، a k (nﻣﻌﻤﻮﻝ ﺍﺳﺖ ﻛﻪ ﻓﺮﺽ ﻛﻨﻴﻢ ﺧﺼﻮﺻﻴﺎﺕ ﺁﻣﺎﺭﻱ ﺳﻴﮕﻨﺎﻝ ﺩﺭ ﻃﻮﻝ ﻳﻚ ﺑﺎﺯﻩ ﻛﻮﺗﺎﻩ ﺯﻣﺎﻧﻲ ﺛﺎﺑﺖ ﻣﻲ ﻣﺎﻧﻨﺪ .ﺿﺮﺍﻳﺐ ﭘﻴﺸﮕﻮ ﺑﻪ ﮔﻮﻧﻪ ﺍﻱ ﺍﻧﺘﺨﺎﺏ ﻣﻲ ﺷﻮﻧﺪ ﺗﺎ ﻣﻴﺎﻧﮕﻴﻦ ﻣﺮﺑﻊ ﺧﻄﺎﻱ ﭘﻴﺸﮕﻮﻳﻲ ﺩﺭ ﻫﺮ ﭘﻨﺠﺮﻩ ﻛﻮﭼﻚ ﺯﻣﺎﻧﻲ ﺣﺪﺍﻗﻞ ﺷﻮﺩ .ﺑﺮﺍﻱ ﺁﺷﻨﺎﻳﻲ ﺑﻴﺸﺘﺮ ﺑﺎ ﻧﺤﻮﺓ ﺍﻧﺘﺨﺎﺏ ﺑﻬﻴﻨﺔ ﺿﺮﺍﻳﺐ ﭘﻴﺸﮕﻮﻳﻲ ﺧﻄﻲ ﺑﻪ ﻣﺮﺟﻊ ] [1ﻣﺮﺍﺟﻌﻪ ﻛﻨﻴﺪ. -3-3اﺳﺘﺎﻧﺪاردﻫﺎي ﮐﺪ ﮐﺮدن ﮔﻔﺘﺎر ﺍﺳﺘﺎﻧﺪﺍﺭﺩﻫﺎﻱ ﺟﻬﺎﻧﻲ ﻣﺘﻌﺪﺩﻱ ﺑﺮﺍﻱ ﻛﺪ ﻛﺮﺩﻥ ﺳﻴﮕﻨﺎﻟﻬﺎﻱ ﮔﻔﺘﺎﺭ ﻭﺟﻮﺩ ﺩﺍﺭﻧﺪ .ﺗﻌﺪﺍﺩﻱ ﺍﺯ ﺍﻳﻦ ﺍﺳﺘﺎﻧﺪﺍﺭﺩﻫﺎ ﺩﺭ ﻟﻴﺴﺖ ﭘﺎﻳﻴﻦ ﺁﻣﺪﻩ ﺍﻧﺪ .ﺑﻪ ﺟﺰ ﺍﺳﺘﺎﻧﺪﺍﺭﺩ G.711ﻫﻤﮕﻦ ﺍﺯ ﻧﻮﻋﻲ ADPCMﺍﺳﺘﻔﺎﺩﻩ ﻣﻲ ﻛﻨﻨﺪ. 10 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 -3آزﻣﺎﯾﺸﺎت (١ﺑﺮﻧﺎﻣﻪ MATLABﻣﻮﺟﻮﺩ ﺩﺭ “demo-quant,” ،Appendixﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺘﻲ ﺭﻭﻱ ﺳﻴﮕﻨﺎﻝ ﺻﻮﺕ ﺍﻧﺠﺎﻡ ﻣﻲ ﺩﻫﺪ. bits ﺑﺮﻧﺎﻣﻪ ﺭﺍ ﺑﺎ ﺩﻗﺖ ﺑﺨﻮﺍﻧﻴﺪ ﺗﺎ ﻣﺘﻮﺟﻪ ﺷﻮﻳﺪ ﭼﮕﻮﻧﻪ ﻛﺎﺭ ﻣﻲ ﻛﻨﺪ .ﺑﺮﻧﺎﻣﻪ ﺭﺍ ﺑﺮﺍﻱ ﻳﻚ ﻓﺎﻳﻞ ﮔﻔﺘﺎﺭ ﺿﺒﻂ ﺷﺪﻩ ﺩﺭ sample bits ١٦ﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ .ﺑﺮﺍﻱ ﻫﺮ ﻣﻮﺭﺩ ،ﺍﻏﺘﺸﺎﺵ ) (Distortionﺩﺭ ﻫﺮ ﺷﻜﻞ ﻣﻮﺝ ،ﻛﻴﻔﻴﺖ ﺻﺪﺍ ﺭﺍ ﺑﺎ ﺗﻐﻴﻴﺮ ﻣﻮﺳﻴﻘﻲ ﺿﺒﻂ ﺷﺪﻩ ﺩﺭ sample ﺳﻄﻮﺡ ﻛﻮﺍﻧﺘﻴﺎﺯﺳﻴﻮﻥ ) (Nﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ .ﺑﺮﺍﻱ ﻫﺮ ﻣﻮﺭﺩ )ﮔﻔﺘﺎﺭ ﻭ ﻣﻮﺳﻴﻘﻲ( N ،ﻣﻮﺭﺩﻧﻴﺎﺯ ﺑﺮﺍﻱ ﺩﺍﺷﺘﻦ ﻳﻚ ﺻﺪﺍﻱ ﺑﺎ ﻛﻴﻔﻴﺖ ﺧﻮﺏ ﻛﺪﺍﻡ ٨ﻭ ﻳﻚ ﻓﺎﻳﻞ ﺍﺳﺖ؟ ﺷﻜﻠﻬﺎﻱ ﺗﻮﻟﻴﺪ ﺷﺪﻩ ﺑﻮﺳﻴﻠﻪ ﺍﻧﺘﺨﺎﺏ ﻫﺎﻱ ﻣﺘﻔﺎﻭﺕ ﺭﺍ ﭘﺮﻳﻨﺖ ﻛﻨﻴﺪ. (٢ﺑﺮﻧﺎﻣﻪ ﺳﻪ ﻧﻤﻮﻧﻪ ﺑﺎﻻ ﺭﺍ ﺑﺮﺍﻱ ﻛﻮﺍﻧﺘﺰﺍﺳﻴﻮﻥ ) m - lawﺑﻪ ﺟﺎﻱ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ( ﺗﻜﺮﺍﺭ ﻛﻨﻴﺪ .ﺷﻤﺎ ﺑﺎﻳﺪ ﻗﺎﺩﺭ ﺑﻪ ﺗﻨﻈﻴﻢ ﭘﺎﺭﺍﻣﺘﺮ mﻋﻼﻭﻩ ﺑﺮ ﺗﻌﺪﺍﺩ ﺳﻄﻮﺡ ﻛﻮﺍﻧﺘﺰﺍﺳﻴﻮﻥ ،N ،ﺑﺎﺷﻴﺪ .ﻧﺘﺎﻳﺞ ﺑﺪﺳﺖ ﺁﻣﺪﻩ ﺑﺎ mﻭ Nﻫﺎﻱ ﻣﺘﻔﺎﻭﺕ ﺭﺍ ﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ .ﺑﺮﺍﻱ ﻳﻚ mﺍﻧﺘﺨﺎﺏ ﺷﺪﻩ ،ﺗﻌﺪﺍﺩ ﺑﻴﺘﻬﺎﻱ ﻻﺯﻡ ﺑﺮﺍﻱ ﺑﺪﺳﺖ ﺁﻭﺭﺩﻥ ﻛﻴﻔﻴﺖ ﻗﺎﺑﻞ ﻗﺒﻮﻟﻲ ﺍﺯ ﮔﻔﺘﺎﺭ ﻭ ﻣﻮﺳﻴﻘﻲ ﭼﻪ ﻣﻲ ﺑﺎﺷﺪ؟ ﺍﻳﻦ ﻣﻘﺎﺩﻳﺮ ﺭﺍ ﺑﺎ ﺑﻴﺖ ﻫﺎﻱ ﻣﻮﺭﺩﻧﻴﺎﺯ ﺩﺭ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ،ﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ. ﺭﺍﻫﻨﻤﺎﻳﻲ :ﺷﻤﺎ ﺑﺎﻳﺪ m - lawﺭﺍ ﺑﻪ ﻣﻘﺪﺍﺭ ﻧﻤﻮﻧﻪ ﺍﻭﻟﻴﻪ ﺍﻋﻤﺎﻝ ﻛﻨﻴﺪ ،ﻣﻘﺪﺍﺭ ﺗﺒﺪﻳﻞ ﻳﺎﻓﺘﻪ ﺭﺍ ﺑﺎ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ،ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻴﺪ ،ﺳﭙﺲ ﻋﻜﺲ m - lawﺭﺍ ﺑﻪ ﻣﻘﺪﺍﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺍﻋﻤﺎﻝ ﻛﻨﻴﺪ ﺗﺎ ﻣﻘﺪﺍﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺩﺭ ﻓﻀﺎﻱ ﺳﻴﮕﻨﺎﻝ ﺍﻭﻟﻴﻪ ﺑﺪﺳﺖ ﺁﻳﺪ .ﺑﺮﺍﻱ ﻋﻜﺲ m - lawﮔﺮﻓﺘﻦ ،ﺷﻤﺎ ﺍﺣﺘﻴﺎﺝ ﺑﻪ ﺗﻌﻴﻴﻦ ) x(nﺍﺯ ) y(nﺩﺍﺭﻳﺪ )ﻣﻌﺎﺩﻟﺔ .(٣-١٣ -٣ﺑﺮﻧﺎﻣﺔ ﻣﻄﻠﺐ ” “sinadm.mﻭ ” “sindm.mﺭﺍ ﺩﺭ Appendixﺑﺨﻮﺍﻧﻴﺪ ﻛﻪ DMﻭ ADMﺭﺍ ﺑﺮﺍﻱ ﻳﻚ ﺳﻴﮕﻨﺎﻝ ﺳﻴﻨﻮﺳﻲ ﭘﻴﺎﺩﻩ ﺳﺎﺯﻱ ﻣﻲ ﻛﻨﺪ .ﻧﺘﺎﻳﺞ ﺑﺪﺳﺖ ﺁﻣﺪﻩ ﺑﺎ dmax, dmin, xmean, p, Qﻫﺎﻱ ﻣﺨﺘﻠﻒ ﺭﺍ ﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ .ﺍﺛﺮﺍﺕ ﺗﻐﻴﻴﺮ Qﺑﻪ ﻣﻘﺎﺩﻳﺮ ﺑﺴﻴﺎﺭ ﻛﻮﭼﻚ ﻭ ﺑﺴﻴﺎﺭ ﺑﺰﺭﮒ ﺭﺍ ﺑﺮﺭﺳﻲ ﻛﻨﻴﺪ .ﻣﺸﺎﺑﻪ ﺁﻥ ﺍﺛﺮﺍﺕ Pﺭﺍ ﻭﻗﺘﻲ ﺑﺴﻴﺎﺭ ﻛﻮﭼﻚ ﻳﺎ ﺑﺰﺭﮒ ﺍﺳﺖ ﻣﻮﺭﺩ ﺑﺮﺭﺳﻲ ﻗﺮﺍﺭ ﺩﻫﻴﺪ .ﻛﺪﺍﻡ ﺩﺳﺘﻪ ﭘﺎﺭﺍﻣﺘﺮﻫﺎ ﺑﻬﺘﺮﻳﻦ ﻧﺘﺎﻳﺞ ﺭﺍ ﺩﺭ ﺍﻳﻦ ﺣﺎﻟﺖ ﻣﻲ ﺩﻫﺪ؟ ﺷﻜﻞ ﻫﺎﻱ ﺗﻮﻟﻴﺪ ﺷﺪﻩ ﺑﺎ ﺍﻧﺘﺨﺎﺏ ﻫﺎﻱ ﻣﺘﻔﺎﻭﺕ ﺭﺍ ﭘﺮﻳﻨﺖ ﻛﻨﻴﺪ. “sindm.m” -٤ﺭﺍ ﺑﻪ ﮔﻮﻧﻪ ﺍﻱ ﺗﻐﻴﻴﺮ ﺩﻫﻴﺪ ﻛﻪ ﺳﻴﮕﻨﺎﻝ ﺻﻮﺕ ﺭﺍ ﭘﺮﺩﺍﺯﺵ ﻛﻨﺪ .ﺑﺮﻧﺎﻣﻪ ﺑﺎﻳﺪ ﻗﺎﺩﺭ ﺑﻪ: ﺍﻟﻒ( ﺧﻮﺍﻧﺪﻥ ﻳﻚ ﻓﺎﻳﻞ ﻭﺭﻭﺩﻱ ﺑﻪ ﻓﺮﻣﺖ .wav ﺏ( ﺍﻋﻤﺎﻝ ﻣﺪﻭﻻﺳﻴﻮﻥ ﺩﻟﺘﺎ ﺑﻪ ﻫﺮ ﻧﻤﻮﻧﻪ ﺝ( ﺑﺎﺯﺳﺎﺯﻱ ﻧﻤﻮﻧﻪ ﺑﻌﺪ ﺍﺯ ﻛﻮﺍﻧﺘﺰﺍﺳﻴﻮﻥ ﺩ( ﻧﻤﺎﻳﺶ ﺷﻜﻞ ﻣﻮﺝ ﺳﻴﮕﻨﺎﻝ ﻫﺎﻱ ﺍﺻﻠﻲ ﻭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ :ﺗﺎ ﺑﻪ ﺷﻤﺎ ﻫﺮ ﺗﻐﻴﻴﺮ ﻛﻮﭼﻜﻲ ﺭﺍ ﺩﺭ ﺩﺍﻣﻨﻪ ﻧﻤﻮﻧﻪ ﻫﺎ ﻧﺸﺎﻥ ﺩﻫﺪ .ﺷﻤﺎ ﺑﺎﻳﺪ ﻳﻚ ﻗﺴﻤﺖ ﻛﻮﭼﻚ ﺍﺯ ﺷﻜﻞ ﻣﻮﺝ ﺭﺍ ﺩﺭ ﻫﻨﮕﺎﻡ ﻧﻤﺎﻳﺶ ﺩﺭ ﻧﻈﺮ ﺑﮕﻴﺮﻳﺪ ﺗﺎ ﻧﻤﻮﻧﻪ ﻫﺎﻱ ﻣﺠﺰﺍ ﺭﺍ ﻭﺍﺿﺢ ﺑﺒﻴﻨﻴﺪ. ﻫـ( ﻓﺎﻳﻞ ﺑﺎﺯﺳﺎﺯﻱ ﺷﺪﻩ ﺭﺍ ﺑﻪ ﻓﺮﻣﺖ ﻓﺎﻳﻞ .wavﺫﺧﻴﺮﻩ ﻛﻨﻴﺪ. ﻭ( ﻓﺎﻳﻞ ﺍﻭﻟﻴﻪ ﻭ ﺑﺎﺯﺳﺎﺯﻱ ﺷﺪﻩ ﺭﺍ ﭘﺨﺶ ﻛﻨﻴﺪ ﺗﺎ ﻛﻴﻔﻴﺖ ﺻﻮﺕ ﺁﻧﻬﺎ ﺭﺍ ﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ. (٥ﺑﺮﻧﺎﻣﻪ ﺭﺍ ﺑﻪ ﺳﻴﮕﻨﺎﻟﻬﺎﻱ ﮔﻔﺘﺎﺭ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺷﺪﻩ ﺩﺭ 11KHzﻭ 22KHzﻭ 44KHzﻭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺑﻪ ٨ﺑﻴﺖ ﺍﻋﻤﺎﻝ ﻛﻨﻴﺪ .ﺑﺮﺍﻱ ﻫﺮ ﻓﺎﻳﻞ ﻭﺭﻭﺩﻱ DM ،ﺭﺍ ﺑﺎ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﺑﺮﻧﺎﻣﺔ MATLABﺍﻋﻤﺎﻝ ﻛﻨﻴﺪ .ﺷﻤﺎ ﺑﺎﻳﺪ ﻃﻮﻝ ﭘﻠﻪ ﺭﺍ ﺑﻪ ﮔﻮﻧﻪ ﺍﻱ ﺗﻨﻈﻴﻢ ﻛﻨﻴﺪ ﻛﻪ ﺑﻬﺘﺮﻳﻦ ﻛﻴﻔﻴﺖ 11 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 ﻣﻤﻜﻦ ﺩﺭ ﻫﺮ ﻣﻮﺭﺩ ﺑﺪﺳﺖ ﺁﻳﺪ .ﺳﻌﻲ ﻛﻨﻴﺪ ﺍﺯ ﻫﻴﺴﺘﻮﮔﺮﺍﻡ ﺍﺧﺘﻼﻑ ﻧﻤﻮﻧﻪ ﻫﺎ ﺑﺮﺍﻱ ﺗﻌﻴﻴﻦ ﻃﻮﻝ ﭘﻠﻪ ﺍﺳﺘﻔﺎﺩﻩ ﻛﻨﻴﺪ .ﻛﻴﻔﻴﺖ ﺻﺪﺍ ﻭ ﺗﻐﻴﻴﺮﺍﺕ ﺩﺍﻣﻨﻪ ﺭﺍ ﺩﺭ ﻫﺮ ﻣﻮﺭﺩ ﻣﺸﺎﻫﺪﻩ ﻛﻨﻴﺪ .ﺩﺭ ﭼﻪ ﻓﺮﻛﺎﻧﺲ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ،ﺳﻴﮕﻨﺎﻝ ﻓﺸﺮﺩﻩ ﺷﺪﺓ DMﻛﻴﻔﻴﺖ ﻗﺎﺑﻞ ﻣﻘﺎﻳﺴﻪ ﺍﻱ ﺑﺎ ﺳﻴﮕﻨﺎﻝ ﺍﺻﻠﻲ ٨ ﺑﻴﺘﻲ ﻭ 11KHzﻓﺮﺍﻫﻢ ﻣﻲ ﻛﻨﺪ؟ ﺑﺮﺍﻱ ﻫﺮ ﻧﺮﺥ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﻓﺮﻛﺎﻧﺴﻬﺎﻱ ﺩﺍﺩﻩ ﺍﺻﻠﻲ ﻭ ﺩﺍﺩﻩ ﺑﻌﺪ ﺍﺯ DMﺭﺍ ﺑﺮﺍﻱ ﻫﺮ ﻧﺮﺥ ﻧﻤﻮﻧﻪ ﺑﺮﺩﺍﺭﻱ ﺑﺪﺳﺖ ﺁﻭﺭﻳﺪ )ﺩﻟﺨﻮﺍﻩ( . bit ﺗﻮﺟﻪ ﻛﻨﻴﺪ ﻛﻪ ﺑﺎ ﺑﺮﻧﺎﻣﻪ ،MATLABﻫﺮ ﭼﻨﺪ ﺧﻄﺎﻱ ﭘﻴﺸﮕﻮﻳﻲ ﺑﻪ sample doubleﺭﺍ ﺩﺍﺭﺩ ﻭ ﻫﻨﮕﺎﻣﻲ ﻛﻪ ﺑﻪ ﻳﻚ ﻓﺎﻳﻞ .wavﺗﺒﺪﻳﻞ ﻣﻲ ﺷﻮﺩ ،ﻫﺮ ﻧﻤﻮﻧﻪ ٨ﻳﺎ ١٦ﺑﻴﺖ ﺟﺎ ﻣﻲ ﮔﻴﺮﺩ .ﺑﻨﺎﺑﺮﺍﻳﻦ ﺍﻧﺪﺍﺯﻩ ﻓﺎﻳﻞ .wavﻛﻪ ١ﻛﻮﺍﻧﺘﻴﺰﻩ ﻣﻲ ﺷﻮﺩ ،ﺳﻴﮕﻨﺎﻝ ﺑﺎﺯﺳﺎﺯﻱ ﺷﺪﻩ ﺩﻗﺖ ﺷﻤﺎ ﺳﺎﺧﺘﻪ ﺍﻳﺪ ،ﻧﻤﺎﻳﺶ ﺩﺭﺳﺘﻲ ﺍﺯ ﺍﻧﺪﺍﺯﻩ ﻓﺎﻳﻞ ﻓﺸﺮﺩﻩ ﺷﺪﻩ ﺣﻘﻴﻘﻲ ﻧﻴﺴﺖ .ﻳﻚ ﺑﺮﻧﺎﻣﻪ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ ﻭﺍﻗﻌﻲ ﺧﻄﺎﻱ ﭘﻴﺸﮕﻮﻳﻲ ﺭﺍ ﺑﺎ bit sample ) (٤ﻭ )(٥ﻭ ) (۶ﺭﺍ ﺑﺎ ﺍﺳﺘﻔﺪﻩ ﺍﺯ ADMﺗﻜﺮﺍﺭ ﻛﻨﻴﺪ .ﺩﺭ ﺍﻳﻦ ﺣﺎﻟﺖ ﺷﻤﺎ ﺑﺎﻳﺪ ﭘﺎﺭﺍﻣﺘﺮﻫﺎﻱ Pﻭ xmeanﻭ dminﻭ dmaxﺭﺍ ﺑﻪ ﻃﻮﺭ ١ﺫﺧﻴﺮﻩ ﻣﻲ ﻛﻨﺪ. ﻣﻨﺎﺳﺐ ﺍﻧﺘﺨﺎﺏ ﻛﻨﻴﺪ .ﺳﻌﻲ ﻛﻨﻴﺪ ﻛﻪ ﺍﺯ ﻫﻴﺴﺘﻮﮔﺮﺍﻡ ﺍﺧﺘﻼﻑ ﻧﻤﻮﻧﻪ ﻫﺎ ﺑﺮﺍﻱ ﺗﻌﻴﻴﻦ ﺍﻳﻦ ﭘﺎﺭﺍﻣﺘﺮﻫﺎ ﺍﺳﺘﻔﺎﺩﻩ ﻛﻨﻴﺪ .ﻛﻴﻔﻴﺖ ADMﺭﺍ ﺑﺎ DM ﺩﺭ ﻛﻮﺍﻧﺘﻴﺰﺍﺳﻴﻮﻥ ﻳﻜﻨﻮﺍﺧﺖ ﻣﻘﺎﻳﺴﻪ ﻛﻨﻴﺪ. (٧ﺍﺧﺘﻴﺎﺭﻱ :ﻳﻚ ﺑﺮﻧﺎﻣﻪ MATLABﺑﺮﺍﻱ ﺗﺤﻘﻖ ﻓﺸﺮﺩﻩ ﺳﺎﺯﻱ" ” ADM + m - lawﺩﺭ ﻳﻚ ﺳﻴﮕﻨﺎﻝ ﻭﺭﻭﺩﻱ ﺩﺍﺩﻩ ﺷﺪﻩ ﺑﻨﻮﻳﺴﻴﺪ. ﺁﻧﺮﺍ ﺑﻪ ﺳﻴﮕﻨﺎﻝ ﻣﻮﺳﻴﻘﻲ ﻛﻪ ﻗﺒﻼﹰ ﺍﺳﺘﻔﺎﺩﻩ ﻛﺮﺩﻩ ﺍﻳﺪ ﺍﻋﻤﺎﻝ ﻛﻨﻴﺪ .ﺁﻳﺎ ﺷﻤﺎ ﺑﻪ ﺑﻴﺖ ﻫﺎﻱ ﻛﻤﺘﺮﻱ ﺑﺮﺍﻱ ﺭﺳﻴﺪﻥ ﺑﻪ ﻫﻤﺎﻥ ﻛﻴﻔﻴﺖ ﺍﺣﺘﻴﺎﺝ ﺩﺍﺭﻳﺪ؟ ﺭﺍﻫﻨﻤﺎﻳﻲ :ﺷﻤﺎ ﺍﺣﺘﻴﺎﺝ ﺑﻪ ﺗﻮﻟﻴﺪ ﺧﻄﺎﻱ ﭘﻴﺸﮕﻮﻳﻲ ﺩﺭ ﻫﺮ ﻧﻤﻮﻧﻪ ﺩﺍﺭﻳﺪ m - law .ﺭﺍ ﺍﻋﻤﺎﻝ ﻛﻨﻴﺪ .ﻣﻘﺪﺍﺭ ﺗﺒﺪﻳﻞ ﺷﺪﻩ ﺭﺍ ﺑﺎ ﺍﺳﺘﻔﺎﺩﻩ ﺍﺯ ﺍﻟﮕﻮﺭﻳﺘﻢ ADMﻛﻪ ﺩﺭ ﺁﻥ ﻣﻘﺪﺍﺭ ﻃﻮﻝ ﭘﻠﻪ ﺗﻌﻴﻴﻦ ﻣﻲ ﺷﻮﺩ ﻛﻮﺍﻧﺘﻴﺰﻩ ﻛﻨﻴﺪ .ﺳﭙﺲ ﻣﻘﺪﺍﺭ ﻛﻮﺍﻧﺘﻴﺰﻩ ﺷﺪﻩ ﺭﺍ ﺑﺎ ﺍﻋﻤﺎﻝ ﻣﻌﻜﻮﺱ m - lawﺑﻪ ﺣﻮﺯﻩ ﺍﺻﻠﻲ ﺑﺮﮔﺮﺩﺍﻧﻴﺪ .ﻣﺎ ﺳﺮﺍﻧﺠﺎﻡ ﺍﻳﻦ ﻣﻘﺪﺍﺭ ﺑﺎﺯﺳﺎﺯﻱ ﺷﺪﻩ ﺭﺍ ﺑﻪ ﻣﻘﺪﺍﺭ ﭘﻴﺸﮕﻮﻳﻲ ﺷﺪﻩ ﺍﺿﺎﻓﻪ ﻣﻲ ﻛﻨﻴﻢ. -5ﮔﺰارش ﺑﺮﻧﺎﻣﻪ ﻫﺎﻱ MATLABﻭ ﺷﻜﻞ ﻫﺎ ) (plotsﺭﺍ ﺗﺤﻮﻳﻞ ﺩﻫﻴﺪ .ﻫﺮ ﭘﺪﻳﺪﻩ ﺍﻱ ﻛﻪ ﻣﺸﺎﻫﺪﻩ ﻛﺮﺩﻩ ﺍﻳﺪ .ﺗﻮﺿﻴﺢ ﺩﻫﻴﺪ .ﺭﻭﻱ ﻛﻴﻔﻴﺖ ﺻﺪﺍ ﺑﺎ ﺗﻨﻈﻴﻢ ﭘﺎﺭﺍﻣﺘﺮﻫﺎﻱ ﻣﺘﻔﺎﻭﺕ ﻧﻈﺮ ﺩﻫﻴﺪ ﻭ ﺳﻮﺍﻻﺕ ﺧﻮﺍﺳﺘﻪ ﺷﺪﻩ ﺩﺭ ﺁﺯﻣﺎﻳﺶ ﺭﺍ ﭘﺎﺳﺦ ﺩﻫﻴﺪ. -6ﻣﺮاﺟﻊ [1]. L.R.Rabiner and R.W.Schafer, Digital Processing of Speech Signals, Prentice Hall 1978 [2]. B.Smith, “Instantaneous Companding of Quantized Signals”, Bell System Tech. J., Vol.36, No.3, pp.653-709, May 1957. [3]. N.S.Jayant, “Adaptive Quantization with a One Word Memory”, Bell System Tech. J., pp. 1119-1144, September 1973. [4]. Guido van. Rossum, “FAQ: Audio File Formats”, http://www.cis.ohio_state.edu. 12 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 13 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 14 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 15 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 16 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 17 CE 342 – Multimedia HW# 2 H. Rabiee, Spring 2008 18
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