參數(shù)資料
型號: PIC24FJ256GB206-I/MR
廠商: Microchip Technology
文件頁數(shù): 20/36頁
文件大?。?/td> 0K
描述: MCU PIC 16BIT FLASH 256K 64VQFN
特色產(chǎn)品: PIC24FJ/33FJ MCUs & dsPIC? DSCs
標準包裝: 40
系列: PIC® 24F
核心處理器: PIC
芯體尺寸: 16-位
速度: 32MHz
連通性: I²C,IrDA,SPI,UART/USART,USB OTG
外圍設(shè)備: 欠壓檢測/復位,LVD,POR,PWM,WDT
輸入/輸出數(shù): 52
程序存儲器容量: 256KB(85.5K x 24)
程序存儲器類型: 閃存
RAM 容量: 96K x 8
電壓 - 電源 (Vcc/Vdd): 2.2 V ~ 3.6 V
數(shù)據(jù)轉(zhuǎn)換器: A/D 16x10b
振蕩器型: 內(nèi)部
工作溫度: -40°C ~ 85°C
封裝/外殼: 64-VFQFN 裸露焊盤
包裝: 托盤
www.microchip.com/16bit
27
Noise Suppression Library
This application library suppresses the noise interference
in a speech signal, such as ambient noise picked up by
a microphone while capturing speech. This algorithm is
particularly useful for systems such as hands-free phones,
speakerphones, intercoms and headsets where an isolated
noise reference is not available. The algorithm handles
0-4 kHz audio bandwidth and provides adjustable noise
reduction up to 44 dB.
Acoustic Echo Cancellation Library
This library provides a function to eliminate the echo
generated in the acoustic path between a speaker and a
microphone, such as in a speakerphone or an intercom
system. This library is compatible with the G.167 standard
and provides 16, 32, 64 or 128 ms echo delays. It handles
0-4 kHz audio bandwidth and provides echo cancellation of
40-50 dB. It also includes some sample rate conversion
functions.
Line Echo Cancellation Library
This library provides a function to cancel electrical line
echoes caused by 2- to 4-wire conversion hybrids in
telephone lines. The library can be used in long distance
voice communication applications, especially in links
involving satellite networks and intercontinental long haul
networks, as well as digital networks, such as Voice over
IP (VoIP). This library is compatible with the ITU-T G.168
recommendation. The library can be used for full-duplex
operation. It handles 0-4 kHz audio bandwidth (8 kHz
sampling of 16-bit speech data).
Equalizer Library
The Equalizer library allows applications to change the tonal
properties of a signal by offering an 8-band frequency band
amplitude control. The library processes an 8 kHz sampled
signal. The library works readily with the rest of Microchip’s
dsPIC DSC Speech and Audio Processing Algorithms.
Libraries for Speech and Audio Applications
Speech Encoding/Decoding Libraries
Three options exist for a variety of speech compression/encoding and decompression/decoding applications:
The library is an implementation of the ITU-T G.711 standard which uses A-law or μ-law
companding to achieve 2: 1 compression.
G.726A is an implementation of the ITU-T G.726 Annex A standard which uses Adaptive Differential Pulse Code
Modulation (ADPCM) encoding algorithm. It can achieve up to an 8:1 compression ratio depending on output bit rate
selected.
Speex is a popular standard in the open source software community which has been adapted for the dsPIC DSC.
It uses Code Excited Linear Prediction (CELP) algorithm. It can achieve a 26:1 compression ratio.
Vocoder
Incoming
Data Rate (16-bit)
Output
Rate
Speech Quality
(MOS)
MIPS(1)
Flash
(KB)
RAM
(KB)
Target
G.711
8 kHz
64 kbps
4.3-4.5
1
3
3.6
PIC24/dsPIC DSC
G.726A
8 kHz
16-40 kbps
4.3-4.5
15
6
4
dsPIC DSC
Speex
8/16 kHz
8-12.8 kbps
3.7-4.2
20/24
24/38
7.6/12
dsPIC DSC
Note 1: Includes both encoder and decoder processing in full duplex.
dsPICDSC Automatic Gain Control Library
The dsPICDSC Automatic Gain Control Library
automatically adjusts the amplitude of a speech signal to
match a set level. This is useful in speech applications
where the distance between the speech source and the
microphone is not fixed. The Automatic Gain Control Library
can be used readily with Microchip’s Speech and Audio
Solutions for speech signal pre-processing.
Speech and Audio Fast Forward (SAFF)
The dsPIC DSC Speech and Audio Fast Forward tool
provides user’s with real-time control of Microchip’s Speech
and Audio Processing Algorithms as they are running in a
embedded system application. This PC GUI tool facilitates
code generation and easy tuning of Microchip’s Noise
Suppression, Acoustic Echo Cancellation, Line Echo
Cancellation and Equalizer algorithms in speech and audio
applications by simplifying parametric tuning of algorithms.
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