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Description: 一个音频采集压缩的VC例子，压缩为AVI格式-an Audio Capture and Compression VC example, compressed AVI format to
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Size: 100352 |
Author: 文心 |
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Description: 根据数据文本转换成二进制文件再写成wave文件的格式转换保存成.wav文件、-According to the text data into binary files and then write wave files into a format conversion to preserve. Wav files,
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Size: 2189312 |
Author: fengyuan |
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Description: 实现8位和16位(*.wav)波型文件的混音,可以是任意多个源文件或波形缓冲区数据.-Realize 8 and 16 (*. wav) file-wave mixing, which can be any number of source files or waveform data buffer.
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Size: 2345984 |
Author: fushou |
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Description: C++实现的读取wave文件的小程序，返回的是声音文件的头地址-this function loads a .wav file, sets up the directsound
buffer and loads the data into memory, the function returns
the id number of the sound
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Size: 2048 |
Author: 丛志丹 |
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Description: Implementation of a speech codec based on coding of speech at 8 kbit/s
using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP)
- We took .wav files that is sampled at 8000 Hz using 16-bit linear PCM. The encoding process is done every 10ms frame or 80 samples. For the preprocessing stage, the samples are high passed with cut-off frequency of 140 Hz and scaled down by 2. A total of 240 samples are buffer for windowing and autocorrelation computation. The autocorrelation coefficients are used to calculate the LP filter coefficients using the Levinson-Durbin algorithm. The LP filter coefficients are converted to Line Spectral Pair (LSP) coefficients. LSP coefficients are converted back to the LP filter coefficients, which is just the reverse process of the conversion from LP to LSP. This module is exactly what the decoder will need in order to convert the LSP coefficients to LP coefficients. We decided not to implement the LSF quantization module because we did not have the codebook information when we designed our system.
The open-loop pitch delay is calculated first for each frame. Then the closed-loop pitch
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Size: 40960 |
Author: coco |
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Description: Wav file is the basic file or raw file which stores the audio data.Wav header contains important information such as sampling rate Monotype or sereo ,bytes per sample.The code reads header and buffer which can be processed the way we want.No additional include file or library files are required.The code is comipled under linux and results were found tobe ok.-Wav file is the basic file or raw file which stores the audio data.Wav header contains important information such as sampling rate Monotype or sereo ,bytes per sample.The code reads header and buffer which can be processed the way we want.No additional include file or library files are required.The code is comipled under linux and results were found tobe ok.
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Size: 2048 |
Author: avinash |
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Description: stc125a60s2 PWM 推动耳机播放SD卡里面的WAV音乐。文件系统为FATFS。内有低层SD卡驱动及双缓冲区的切换。-stc125a60s2 PWM promote headphone playback SD card inside WAV music. The file system FATFS. Within the lower the SD card drive and double buffer switching.
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Size: 35840 |
Author: ethan |
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Description: 这只是一段简短的程序 把文件读到buffer中-load wave to buffer
Platform: |
Size: 1024 |
Author: Yangquan Ni |
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Description: winmm.dll中的mciSendString方法录音C#代码
mciSendString是用来播放多媒体文件的API指令，可以播放MPEG,AVI,WAV,MP3,等等
该函数有四个参数：
第一个参数：要发送的命令字符串。字符串结构是:[命令][设备别名][命令参数].
第二个参数：返回信息的缓冲区,为一指定了大小的字符串变量.
第三个参数：缓冲区的大小,就是字符变量的长度.
第四个参数：回调方式，一般设为零-the winmm.dll of the mciSendString method of recording the C# code mciSendString API instruction is used to play multimedia files that can play MPEG, AVI, WAV, MP3, etc. The function has four parameters: the first parameter: To send a command character string. String structure is: [command] [device alias] [command parameters]. Second parameter: Returns information buffer size for a specified string variable. Third parameter: the size of the buffer is character variable length. fourth parameter: callback, generally set to zero
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Size: 1024 |
Author: 刚 |
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