搜索资源列表
IPphone2.0
- 2.0说明 1.0支持系统的录音多种格式,有一定的延时,没有过滤杂音功能. 2.0已取消支持系统的录音多种格式,使用PCM采集数据G711A压缩格式(8000HZ单声道16位格式录音每秒以8K完成数据,16000HZ单声道16位格式录音每秒以11K完成数据,音质相当好),延时降低到最小100-500MS以内,不会随时间增加而增加延时(如果是说话测试一直保持200MS的延时,如果是用播放歌曲来测试,有自动校正延时功能,恢复成200MS的延时,恢复过程中不会中断歌曲的播放,只是小小加快唱歌的
AudioPcm(Mu-Law)Source20061215
- DirectShow开发的PCM Mu-Law的音频解码Filter,C-DirectShow development of the PCM Mu-Law Audio Decoder Filter, C
AudioPcm(Raw-Signed)Source20061206
- DirectShow开发的PCM Raw Signed 8K 8Bit音频解码Filter,VC++6.0-DirectShow development of the PCM 8K Raw Signed F 8Bit Audio Decoder ilter, VC 6.0
PCMSource
- PCM音频编码Source filter-PCM Source Filter
WavDest
- DirectShow实现对音频文件到PCM的转换Filter-DirectShow audio files to PCM to realize the conversion Filter
encoder
- 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 i
PCM
- PCM信号的码同步提取；短脉冲滤除；VHDL语言-PCM code synchronization signal extraction short pulse filter VHDL language
voice2.0
- 基于stm32的数字化语音存储与回放系统的软件设计。该系统经过拾音器采集语音信号，经带通滤波与AGC后，进入AD，然后数据存入SRAM，再由DA送至带通滤波，由喇叭将语音信号恢复。该系统可以很好的存储与回放语音信号。， 本源码是在mdk上开发的。 软件设计分两个模块：压缩解压编码与存储模块。压缩编码有PCM编码和ADPCM编码两种可用按键选择；存储是FSMC存储（256k flash）存储。-Stm32 of digital voice storage and playback system
RSC
- RSC实现人脸识别，使用PCM降维并且采用gabor滤波器来提高识别率-RSC face recognition, using the PCM dimensionality reduction and the use of gabor filter to improve the recognition rate
yinyue
- 单片机读取SD卡音乐直接利用SPWM波滤播放，在播放前需要将音频文件转化为PCM格式。-Microcontroller reads the SD card music playback directly use the SPWM wave filter in front of playing audio files need to be converted to PCM format.
ControlVolume
- 通过filter控制PCM音频音量的大小 输入输出PCM格式-The filter used to control the volume of the audio
SP-kalman_Matlab
- kalman滤波器经典入门matlab程序 [Simulink_kalman.rar] - 实现kalman滤波算法，通过跟踪，估计物体运动轨迹。-,matlab,matlab例程/matlab，做了仿真 [matlab.rar] - 收集的matlab，simulink的全部基本命令，方便大家查阅！ [selfadptivesignalprocess.rar] - 自适应信号处理算法的实现，非常有价值 [pcm.rar] - 语音编码方案的选取对移动通信系统的通话质量
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.
NdkMediaError
- Apply FIR filter to 44 kHz raw 16-bit PCM linear audio then downsample to 11 kHz.
filter.CompressDTS
- 把PCM音频流保存为DTS文件的dshow过滤器；-Save the PCM audio stream to DTS file dshow filter