Description: 本程序是基于ACELP算法编写的语音编解码程序，里面有两个程序例子，一个是编码程序，另一个是解码程序。-This is a speech codec based on ACELP algorithm,there are two examples in it,one is the encoder,anther is decoder. Platform: |
Size: 100454 |
Author:吴中平 |
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Description: AMR speech encoder and decoder ETSI Release 7.4.0-AMR speech encoder and decoder ETSI Releas e 7.4.0 Platform: |
Size: 803332 |
Author:朱平 |
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Description: va_g729a.lib Win32 statically linkable library of G729 floating-point object code for Pentium and compatible processors.
va_g729a.h API prototypes and constants declarations required by the sample programs.
va_g729a_encoder.c
Encoder sample application demonstrating encoder API calls to the codec for encoding a speech file.
va_g729a_decoder.c
Decoder sample application demonstrating decoder API calls to the codec for decoding a speech file.
va_g729a_encoder.exe
Encoder sample program executable for the Win32 platform.
va_g729a_decoder.exe
Decoder sample program executable for the Win32 platform. Platform: |
Size: 362822 |
Author:ycyhjj2865@126.com |
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Description: 本程序是基于ACELP算法编写的语音编解码程序，里面有两个程序例子，一个是编码程序，另一个是解码程序。-This is a speech codec based on ACELP algorithm,there are two examples in it,one is the encoder,anther is decoder. Platform: |
Size: 2482176 |
Author:吴中平 |
Hits:
Description: AMR speech encoder and decoder ETSI Release 7.4.0-AMR speech encoder and decoder ETSI Releas e 7.4.0 Platform: |
Size: 802816 |
Author:朱平 |
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Description: AMR语音编码器算法,26204-710修正了以前版本的问题.其中decoder为定点版本.encoder为浮点版本-AMR Speech Coding Algorithm ,26204-710 amended the previous version of the problem. One decoder for the fixed-point version. Encoder for the floating-point version Platform: |
Size: 261120 |
Author:LI JIANZHONG |
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Description: Simulation of Residual Excited Linear Prediction (RELP) coding for speech:
This simulation give your voice or available clear wav file.This encoder have linear predictor that decreases signal s dynamic (lower quantization level). This technique also use lowpass filtering to make lower bandwidth and reconstruct it with three methods that is chosen in decoder. For example duplicate same spectrum of encoded signal that filtered in upper frequency.-Simulation of Residual Excited Linear Prediction (RELP) coding for speech:
This simulation give your voice or available clear wav file.This encoder have linear predictor that decreases signal s dynamic (lower quantization level). This technique also use lowpass filtering to make lower bandwidth and reconstruct it with three methods that is chosen in decoder. For example duplicate same spectrum of encoded signal that filtered in upper frequency. Platform: |
Size: 855040 |
Author:Ardalan |
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Description: it is code for gsm speech coing
in the main.m,encoder.m,decoder.m,levision.m m-file .
it is helping in implement on matlab platform Platform: |
Size: 2048 |
Author:Viral |
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Description: ITU-T G.722.2
国际电信联盟G.722.2建议书，2003年7月版。该建议书是语音通讯领域的压缩标准，被GSM，WCDMA，3GPP等采用，题目为16kbit下的宽带语音编码，使用自适应多率宽带编码。
内容主要有代数码激励线性预测编码(ACELP)，话音活动检测(VAD)等。-This Recommendation describes the high quality Adaptive Multi-Rate Wideband (AMR-WB)
encoder and decoder that is primarily intended for 7 kHz bandwidth speech signals. AMR-WB
operates at a multitude of bit rates ranging from 6.6 kbit/s to 23.85 kbit/s. The bit rate may be
changed at any 20-ms frame boundary.
Annex C includes an integrated C source code software package which contains the implementation
of the G.722.2 encoder and decoder and its Annexes A and B and Appendix I. A set of digital test
vectors for developers is provided in Annex D. These test vectors are a verification tool providing an
indication of success in implementing this codec.
G.722.2 AMR-WB is the same codec as the 3GPP AMR-WB. The corresponding 3GPP
specifications are TS 26.190 for the speech codec and TS 26.194 for the Voice Activity Detector. Platform: |
Size: 728064 |
Author:刘涛 |
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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 Platform: |
Size: 40960 |
Author:coco |
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Description: huffman
哈夫曼编/译码器
利用哈夫曼编码进行通信可以大大提高1言道利用率，缩短信息传速时间，降低传输成本。但是．这要求在发送端通过一个编码系统对待传数据预先编码．在接收端将传来的数据进行译码（复原）。对于双工信道（即可以双向传输俏息的信道），每端都需要一个完整的编/译码系统。试为这样的信息收发站写一个哈夫曼码的编／译码系统。
-huffman Huffman encoder/decoder to communicate using Huffman coding can greatly improve a speech channel utilization, reduce the speed of information transmission time and reduce transport costs. However. This requires the sender to pass through a coding system to treat the data pre-encoding. At the receiving end to decode data from the (recovery). For the duplex channel (ie, interest rates can be pretty two-way transmission channel), each side needs a complete encoding/decoding system. Test for such messaging station to write a Huffman encoding/decoding system. Platform: |
Size: 2048 |
Author:尘封 |
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Description: 语音编码算法实现论文,可以让你边了解dsp原理，边学算法，一举两得啊-Papers speech coding algorithm, can you understand the side dsp principle, learning algorithm, kill two birds with one stone ah! ! ! Platform: |
Size: 907264 |
Author:朱仁安 |
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Description: Watermarking is a process in which both physical
and digital media are marked using watermarks in order to
protect ownership of the watermarked media. Digital water-
marking is a technique where a watermark gets embedded into
the carrier signal while preserving the quality of the original
media. Embedding can happen in various domains and could
be both hidden and plain, but the quality and the information
carried by the signal should not deteriorate. This paper deals
with hiding watermarks into speech audio signals using deep
neural networks. We present an encoder-decoder architecture
that achieved PSNR value greater than 57dB, which we used
as a preservation measure of the original signal and message
transmission accuracy of almost 100%. Audio data, used in this
paper, consists of speeches from the Parliament of Montenegro. Platform: |
Size: 1057925 |
Author:bamzi334 |
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Search - speech encoder decoder