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LAB
- 从均匀PCM方法入手，产生一振幅为1和频率ω=1的正弦信号，分别用8电平和16电平进行量化，在同一坐标轴上绘出原信号和以量化信号，并比较这两种情况下的信号/量化噪声比-Starting from the uniform PCM method to produce a 1 amplitude and frequency ω = 1 of the sinusoidal signal, respectively, with 8 levels and 16 levels were quantified
FPGAMP3_LUKA_Project_Proposal
- The goal of this project is to design a MPEG Layer III (MP3) player using a FPGA board. The FPGA will read MP3 source files, decode them into a 16-bit Pulse Code Modulated (PCM) output, and play the audio files through an external speaker.
ccs_c_manual
- C Compiler Reference Manual.The PCB, PCM, and PCH are separate compilers. PCB is for 12-bit opcodes, PCM is for 14-bit opcodes, and PCH is for 16-bit opcode PIC microcontrollers.
G711G721G723
- G.711算法，它是国际电信联盟ITU-T订定出来的一套语音压缩标准，它代表了对数PCM（logarithmic pulse-code modulation）抽样标准，主要用于电话。它主要用脉冲编码调制对音频采样，采样率为8k每秒。它利用一个 64Kbps 未压缩通道传输语音讯号。 起压缩率为1：2， 即把16位数据压缩成8位。G.711是主流的波形声音编解码器。-G.711 algorithm
Wav_play_echo
- The following classes, CRecordSound and CPlaySound, record sound and play PCM sound simultaneously. An example dialog-based program is provided that records, saves to disk and then echos sound. This example also contains two more classes, CWriteSound
ccs_c_manual
- The PCB, PCM, and PCH are separate compilers. PCB is for 12-bit opcodes, PCM is for 14-bit opcodes, and PCH is for 16-bit opcode PIC® microcontrollers. Due to many similarities, all three compilers are covered in this reference manual. Features a
U_PCM_matlab
- 产生一个正弦信号，用均匀PCM方法分别用8电平和16电平做量化，在同一坐标轴画出原信号和已量化信号，比较两种情况下的SQNR-the matlab source code of U_pcm
exp1
- 提交一段录制语音，录音内容 : 数字0~9 和 本人姓名。格式为单声道、16KHz 采样、 16Bits 的线性PCM 数据。录音要求发音平稳，匀速，音量适中（最大幅度要超过3000）。录 音数据按照WAVE 文件格式存放。文件名由和姓名组成: 学号_姓名.wav 1）编写程序(MathLab)，读出所录制的数据。统计文件信号的直流分量，消除直流后显示波 形。 2）编写程序（MathLab）以100 帧/秒的频度（即帧移10mS），分析并显示录音文件的短时 过零率及短时能
pcm2wav
- PCM数据生成WAV文件的工具，PCM数据为8000采样率16位单声道-PCM data to WAV file, PCM data must be 8000 samplerate, 16 bits and mono
MicrophoneInputStream
- PCM input stream from the microphone, 16 bits per sample.
volumegain
- 16位PCM格式音量自动增益 调节参数可放大 缩小音量-16bit and PCM formation volume auto gain
code
- ADPCM解码器,4位adpcm音频数据解压缩成16位的pcm数据，采样频率为20KHz.-ADPCM decoder
echo
- speex中开源回声消除程序，可以直接运行模拟回声消除程序，针对于VOIP中的回声消除，输入和参考信号都是16位的PCM码-speex echo cancellation in the open-source program, you can run the simulation program directly echo cancellation for echo on VOIP in the elimination of the input and reference signals are
SPEECH_DCODE2
- 这是一个关于ADPCM解压算法，只要给一个ADPCM值就可以输出16位PCM码-This is a ADPCM decompression algorithm, as long as a ADPCM value of 16 can output PCM code
wifi_audio_app
- 基于TI CC3200-LAUNCHXL开发板和CC3200AUDBOOST的WIFI音频对讲DEMO程序，提供的音频演示应用需要两个基站模式的 C3200-LAUNCHXL + CC3200AUDBOOST 设置，并通过 Wi-Fi 向第二个LaunchPad 提供 16 kHz、16 位采样的 PCM 音频的单播流-Based on TI CC3200-LAUNCHXL WIFI Development Boards CC3200AUDBOOST of DEMO program audio
file
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
filter
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
init
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
ip_conntrack
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
narrow_many
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.